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 TROUBLESHOOTING GUIDE

OmniPCX Enterprise

No. TG00069

Ed. 07

Nb of pages :141

Date : 9th January 2013

SUBJECT : Session Initiation Protcol (SIP)

CONTENTS

1.

INTRODUCTION
..........................................................
6

2.

DOCUMENT HISTORY
..........................................................
6

3.

REFERENCES
..........................................................
6

4.

ABBREVIATIONS AND
NOTATIONS............................................. 6

4.1

Abbrevations
..........................................................
6

4.2

Notations
..........................................................
6

PROTOCOL
..........................................................
7

5.

5.1

SIP Overview
..........................................................
7

5.2

SIP Terminology
..........................................................
7

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5.3

SIP structure
..........................................................
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8

5.4

SIP Messages
..........................................................
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8

5.5

SIP Transaction, Dialog & Session

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..........................................................
9

5.5.1

Transaction
..........................................................
9

5.5.2

Dialog
..........................................................
10

5.5.3

Session
..........................................................
10

5.6

SIP Addressing
..........................................................
10

6.

SIP LICENSING
..........................................................
11

7.

SIP OXE IMPLEMENTATION

......................................................
12

7.1

RFCs implemented on OXE

..........................................................
12

7.1.1

SIP
..........................................................
12

7.1.2

RTP, T38 & DTMF (used for SIP)

..........................................................
13

7.2

SIPMOTOR processes
..........................................................
14

7.3

OXE duplication
..........................................................
14

7.4

The OXE contains the following compoments:

 ........................................ 15

7.4.1

Registrar.................................................
15

7.4.2

Proxy
..........................................................
15

7.4.3

Gateway...................................................
17

7.4.4

Dictionnary
..........................................................
17

7.4.5

SIP users
..........................................................
17

7.4.6

SIP External Voice Mail

..........................................................
18

7.5

Overview of Interactions between Components

...................................... 18

7.6

Network number rules

..........................................................
19

7.7

SIP parameters explanation / under the object SIP:

................................ 19

7.7.1

SIP Trunk Group

..........................................................
19

7.7.2

The local SIP gateway

..........................................................
21

7.7.3

The external SIP gateways

..........................................................
22

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7.7.4

Timer usage for SIP Trunking (Trunk Categoy, by default

31).......................... 24

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7.7.5

The SIP proxy

..........................................................
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24

7.7.6

SIP Registrar
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..........................................................
25

7.7.7

SIP Dictionnary
..........................................................
25

7.7.8

SIP Authentication
..........................................................
26

7.7.9

Quarantined IP Addresses
..........................................................
26

7.7.10

Trusted IP Addresses
..........................................................
26

7.7.11

SIP To CH Error Mapping

..........................................................
26

7.7.12

CH To SIP Error Mapping

..........................................................
28

7.8

SIP parameters explanation / under the object USERS:

........................... 28

7.8.1

SIP Device
..........................................................
28

7.8.2

SIP Extension (or SEPLOS)

..........................................................
29

7.9

SIP parameters explanation / under the object SIP

Extension: ................. 29

7.10 SIP parameter explanation / under the object

External Voice Mail: .......... 30

7.11 SIP parameters explanation / under the object

System:........................... 31

IP DOMAINS, CODECS AND PCS

............................................... 32

8.

8.1

IP domains rules
..........................................................
 32

8.2

System law for PCM codec

..........................................................
32

8.3

Codecs on SDP
..........................................................
32

8.3.1

Initial offer : the offer sent in an initial

INVITE................................................
32

8.3.1

Initial answer : the answer to an initial offer on

incoming call ....................... 33

8.4

PCS
..........................................................
33

CONTENTS OF A SIP MESSAGES (GENERAL VIEW)

.................. 34

9.

9.1

The HEADER
..........................................................
34

9.2

The BODY
..........................................................
36

10. EXAMPLES OF COMMON SIP FLOWS

....................................... 38

10.1 Registration
..........................................................
38

10.2 De-registration
..........................................................
41

10.3 Simple call establishement

..........................................................
42

11. TROUBLESHOOTING
..........................................................
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46

11.1 SIPMOTOR processes

..........................................................
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46

11.2 SIPMOTOR memory used

..........................................................
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47

11.3 Check the SYSTEM and SIPMOTOR backtraces/alarms

............................ 48

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11.3.1

Backtraces
..........................................................
48

11.3.2

Alarms
..........................................................
49

11.4 SIP
traces....................................................
51

11.4.1

SIPMOTOR traces
..........................................................
51

11.4.2

Call Handling
traces....................................................
53

11.4.3

Tcpdump / Network traces

..........................................................
54

11.5 Mantenance commands

..........................................................
55

11.5.1

sip
..........................................................
55

11.5.2

trkstat
..........................................................
55

11.5.3

trkvisu
..........................................................
56

11.5.4

sipaccess
..........................................................
57

11.5.5

sipgateway
..........................................................
57

11.5.6

sipdump
..........................................................
58

 11.5.7

sipextgw
..........................................................
67

11.5.8

sippool
..........................................................
68

11.5.9

sipdict
..........................................................
69

11.5.10

sipauth
..........................................................
70

11.5.11

sipregister
..........................................................
71

11.5.12

csipsets
..........................................................
72

11.5.13

csipview com
..........................................................
73

11.5.14

csiprestart
..........................................................
74

11.5.15

sipextusers (Only in R10.x for Open Touch).

................................................ 74

11.6 Link between SIPMOTOR traces and Call Handling

traces ....................... 75

11.6.1

Call Handling / SIPMOTOR links implementation

........................................ 75

11.6.2

General view
..........................................................
76

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11.6.3

“neqt” link between SIPMOTOR and Call Handling traces

.......................... 77

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11.7 Information on the SIPMOTOR traces
..................................................... 78

11.8 Follow a call on the SIPMOTOR trace

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..................................................... 78

11.9 Traces analyses

..........................................................
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81

11.9.1

Incoming SIP call using a SIP Trunk Group: SIPMOTOR

point of view ............ 81

11.9.2

Incoming SIP call using a SIP Trunk Group: Call Handling

point of view ......... 91

11.9.3

Incoming SIP call in case of SIP extension: SIPMOTOR

point of view ............. 97

11.9.4

Incoming SIP call in case of SIP extension: Call

Handling point of view ........ 108

11.10 Main call flows explanation

..........................................................
115

11.10.1

Forwards
..........................................................
115

11.10.2

Transfer
..........................................................
117

11.10.3

UPDATE on Early Media

..........................................................
121

11.11 Configuration issues

..........................................................
123

11.11.1

SIP configuration rule

..........................................................
123

11.11.2

SIP alarms generated on

OXE.......................................................
124

11.11.3

Common SIP issues

..........................................................
126

11.11.4

SIP Device issues

..........................................................
130

11.11.5

SIP extension issues

..........................................................
 131

11.11.6

SIP External Gateway

Issue.....................................................
132

11.12 Use case

..........................................................
133

11.12.1

Outgoing Call – Cancel sent by OXE after 180 w SDP

............................... 133

11.12.2

Telephone-event are not provided on SDP offer

........................................ 133

11.12.3

Loss of communication with SIP External Voicemail

................................... 133

11.12.4

Impossible to let a message when routing via SIP

Automated Attendant... 133

11.12.5 When call is transfer from a Third Party Server,

after few seconds, a Re-Invite

is sent by OXE to reroute RTP to a GD card

..........................................................
133

11.12.6 Incoming call from a SIP Third Party Server is

rejected by OXE with a SIP Error

488 Not Acceptable Here

..........................................................
133

11.12.7

Incoming call is not recognized as INTERNATIONAL

................................. 134

11.12.8 When we attempt to register on SIP External

Gateway, OXE answers by a SIP

error “482 Loop Detected”

..........................................................
134

11.12.9 When we attempt to register our SIP External

Gateway with an external SIP

Proxy, SIP Proxy answers by a SIP error “416 Unsupported

URI Scheme” .................. 135

11.12.10

Incoming call doesn’t transit via Trunk Group configured

on SIP Ext Gw 135

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11.12.11

Wrong caller number sent in case of forward

........................................ 136

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11.12.12

Diversion/History-Info header is not

present.......................................... 136

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11.12.13

SIP-Trunking Name is displayed on calling phone set when

call is

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established 137

11.12.14

From header has not the national format

.............................................. 137

11.12.15

Incoming and outgoing fax communications impossible

through SIP Gw137

11.12.16

No Re-Invite with T38 offer sent by OXE

................................................ 137

11.12.17

External call with secret identity over SIP Provider

fails .......................... 137

11.12.18

On SIP outgoing call, dynamic ports are used instead of

port 5060 ....... 138

11.12.19

A "+" character is added on calling number when ISDN

call is routed to SIP

138

11.12.20

Diversion Field has not the canonical form

............................................ 138

11.12.21

Leg1 and leg2 are external set, when OXE user performs a
blind transfer, it

doesn’t work
..........................................................
139

11.12.22

SingleStep Transfer with REFER, no referred-by in the

following INVITE 139

11.13 Summary for SIP issue analyse

..........................................................
140

BEFORE CALLING ALCATEL-LUCENT’S SUPPORT CENTER

............... 141

NOTE
..........................................................
141

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

1. INTRODUCTION

This Troubleshooting Guide deals with SIP (Session

Initiation Protocol) and its implementation in

OmniPCX Enterprise (OXE), which allows the OXE to

connect to SIP phones, SIP trunks and

 SIP applications like external Voicemail.

The goal is of this document is to explain the

functioning of the SIP, to facilitate the
troubleshooting

and resolution of issues related to SIP

2. DOCUMENT HISTORY

Ed01: first edition

Ed02: add “Traces analyses” chapter

Ed03: add “Use Case” chapter and update 7.11 section

Ed04: update “SIP Device issues” chapter

Ed05: update “Use Case” chapter

Ed06: update 7.7.3 chapter, add new chaper “Timer Usage

for SIP Trunking”

Ed07: add Restriction on “Support of Re-Invite wo SDP”,

see 7.7.3 chapter

3. REFERENCES

OmniPCX Enterprise Technical Documentation

4. ABBREVIATIONS AND NOTATIONS

4.1

4.2

Abbrevations

OXE

: OmniPCX Enterprise

SIP

: Session Initiation Protocol

URI

: Uniform Resource Identifier

Notations

We suggest to pay attention to this symbol, which

indicates some possible risks or gives important

information.

Ed. 07

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

5. PROTOCOL

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5.1

SIP Overview

The SIP protocol is designed to establish, to maintain

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and to end multimedia sessions between different

parties. This protocol is based on the HTTP 1.1

SIP does not provide an integrated communication system.

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SIP is only in charge of initiating a dialog

between interlocutors and of negotiating communication

parameters, in particular those concerning the

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media involved (audio, video). Media characteristics are
described by the Session Description Protocol

(SDP). SIP uses the other standard communication

protocols on IP: for example, for voice channels on IP,

Real-time Transport Protocol (RTP) and Real-time

Transport Control Protocol (RTCP). In turn, RTP uses

G7xx audio codecs for voice coding and compression.

Application Layer

Transport Layer

SDP

MEDIA CODING

SIP

RTP/RTCP

TCP

UDP

Network Layer

5.2

IP

SIP Terminology

User Agent (UA)

User Agent Client (UAC): Initiator of the SIP requests

User Agent Server (UAS): Receiver of the SIP requests

(end point)

A SIP equipment can be UAC or UAS according to the

direction of the call

Call Direction

Alice

Bob

UAC

UAS

Call Direction

Alice

Bob

UAS

UAC

Registrar: A registrar is a server that accepts REGISTER

requests and places the information it

receives in those requests into the location service for

the domain it handles.

The OmniPCX Enterprise incorporates the function of

registrar.

Location Service: A location service is used by a SIP

redirect or proxy server to obtain information

about a callee's possible location(s). It contains a

list of bindings of address-of-record keys to zero

or more contact addresses.

The OmniPCX Enterprise incorporates the function of

location service.

 Proxy, Proxy Server: An intermediary entity that acts as

both a server and a client for the purpose of

making requests on behalf of other clients. A proxy

server primarily plays the role of routing, which

means its job is to ensure that a request is sent to

another entity "closer" to the targeted user.

Ed. 07

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Proxies are also useful for enforcing policy (for

example, making sure a user is allowed to make a

call). A proxy interprets, and, if necessary, rewrites

specific parts of a request message before

forwarding it. The SIP proxy is the central actor and

first contact for any SIP end user device that

wants to initiate a request.

Note: In the OmniPCX Enterprise, the logical functions

of registrar, location service and proxy server

are co-located and running on the OmniPCX Enterprise

call server (CPU/CS/AS) board. The

OmniPCX Enterprise proxy server is stateful (it

remembers transaction state), call-stateful (stays in

the signaling path) and forking (it can redirect

requests to multiple destinations).

The name of the SIP domain handled by an OXE node is its

node name concatenated with the DNS

local domain name defined in SIP/SIP gateway. The main

IP address can be substituted wherever

appropriate.

Redirect Server: Provides the client with information

about the next hop or hops that a message

should take and then the client contacts the next hop
server or UAS directly.

OmniPCX Enterprise does NOT provide a redirect server.

Gateway: A gateway is a SIP user agent that provides a

bridging function between the SIP world and

other signaling and telephony systems.

5.3

SIP structure

The SIP is based on the RFC 3261 (previous RFC 2543),

its implementation is next:

Application

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Transaction user

Transaction

Transport

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5.4

Session, dialog

Traitement of the services

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Treatment, retransmission of messages

Emission, reception of the messages

Syntax/Encoding

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Analyse of the messages (Parsing)

UDP

Transport protocol

TCP

SIP Messages

The main types of requests are:

REGISTER: message sent by an agent to indicate his

current address. This information can be

stored in the location server and is used for call

routing.

INVITE: message sent systematically by the client for

any connection request.

ACK: message sent by the client to confirm (acknowledge)

the connection request.

BYE: terminates a call, RTP packet exchange is stopped.

CANCEL: terminates a call currently being set up.

SUBSCRIBE - NOTIFY: message used to subscribe to/notify

an event (for example: new voicemail

message).

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

REFER: message requesting an agent to call an address

(used for transfers).

UPDATE: message sent to change the SDP information in

early dialog or confirmed dialog.

MESSAGE: message used to send a message.

OPTIONS: Requests information about the capabilities of

a caller, without setting up a call. Also

used for supervision purpose between two Uas.

PRACK: (Provisional Response Acknowledgement): PRACK

improves network reliability by adding

an acknowledgement system to the provisional Responses

(1xx). PRACK is sent in response to

provisional response (1xx).

The remote endpoint answers with a response of one of

the following types (main messages answered by

OXE):

1xx: informational (transaction in progress).

The 100 Tyring is particular regarding the other

informational answers, used to avoid

retransmission of INVITE.

The 180 Ringing is used for ring back tone (RBT).

The 183 Progress is used to broadcast voice guides.

 2xx: success (transaction completed successfully).

200 Ok indicates the request was successfull

202 Accepted indicates that the request has been

accepted for processing, but the

processing has not been completed

3xx: forward (the transaction is terminated and prompts

the user to try again in other conditions).

301 Moved Permanently

302 Moved Temporarily

4xx: The request contains bad syntax or cannot be

fulfilled at the server.

5xx: The server failed to fulfill an apparently valid

request

6xx: The request cannot be fulfilled at any server

Regarding the unsuccessfull answers, for signification,

use the RFC 3261.

5.5

SIP Transaction, Dialog & Session

5.5.1

Transaction

The transactions have to separated:

The INVITE transaction

The INVITE transaction is composed of three ways

INVITE sends from the client to the server

Answers send from the server to the client

Client must send an ACK

If these three steps are respected, a INVITE transaction

is done

Example

UAC

Ed. 07

UAS

INVITE

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OmniPCX Enterprise

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

|--------------->|

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
|

100 Trying |

|<---------------|

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|

180 Ringing |

|<---------------|

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
|

200 OK

|<---------------|

ACK

|--------------->|

An INVITE transaction (with all the information from

this INVITE) can be called a “leg”.

The Non-INVITE transaction

The Non-INVITE transaction is composed of two ways

Request sends from the client to the server

Answers send from the server to the client

No ACK

If these three steps are respected, a Non-INVITE

transaction is done

Example

UAC

UAS

Option

|--------------->|

200 OK

|<---------------|

5.5.2

Dialog

Dialogs are created through the generation of non-

failure responses. When an INVITE is answered with a

200 Ok, the dialog is opened.

A dialog is identified by :

o a call identifier

o a local tag

o a remote tag

5.5.3

Session

A session is open for audio or video exchanges. The UAC

and UAS receives the information to open a RTP

flow, in that case, the session is opened.

5.6

SIP Addressing

SIP entities are identified using SIP URIs (Uniform

Resource Identifier). A SIP URI is of the form of

sip:username@host, similar to an email address,

 typically containing a username and a host name
delimited

by @ (at) character. The host part can be an IP address,

the name of a machine, or a Fully Qualified Domain

Name (FQDN), i.e. the name of a domain. The username

part can be a telephone number.

Examples for SIP URIs:

sip:[email protected]

sip:[email protected]

sip:[email protected]

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

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In OmniPCX Enterprise, the more specific term URL

(Uniform Resource Locator) is generally used instead of

URI, since OXE is more concerned about location aspects

rather than identification aspects.

For OXE uses on the username part numbers and no names.

6. SIP LICENSING

Here the next licenses for SIP (under spadmin):

177 M SIP users

...

185

SIP Gateway

...

188

SIP network links

...

345 M SIP extension users

13/ 25

45

8/ 25

The license 177 corresponds to the maximum number of SIP

users (SIP Extension & SIP Device).

The license 185 corresponds to the use of the SIP on the

OXE (activation).

The license 188 corresponds to the maximum number of SIP

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Calls available all the SIP elements

(SIP calls thru Trunk group and SIP extension).

The license 245 corresponds to the maximum number of SIP

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Extension users.

Another information link to SIP is important, the

PARAMAO 3 used for the creation of the SIP Trunk Group

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(under cfgUpdate):

5 Trunks

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5000

This value is calculated according to the number of

Trunk Groups managed via ACTIS (including SIP).

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

7. SIP OXE IMPLEMENTATION

7.1

RFCs implemented on OXE

7.1.1

SIP

RFC 2543 (obsolete by RFC 3261,3262, 3263,3264, 3265):

SIP: Session Initiation Protocol

RFC 2782: A DNS RR for specifying the location of

services (DNS SRV)

RFC 2822: Internet Message Format

RFC 3261: SIP: Session Initiation Protocol

RFC 3262: Reliability of Provisional Responses in SIP

(PRACK)

RFC 3263: SIP: Locating SIP Servers

RFC 3264: An Offer / Answer model with SDP

RFC 3265: SIP-Specific Event Notification

RFC 3311: The SIP UPDATE Method (session timer only)

RFC 3323: Privacy Mechanism for the Session Initiation

Protocol (SIP)

RFC 3324: Short term requirements for network asserted

identity

RFC 3325:Private Extensions to the Session Initiation

Protocol (SIP) for Asserted Identity within

Trusted Networks

RFC 3265: SIP-specific Event Notification

RFC 3515: The Session Initiation Protocol (SIP) Refer

method

RFC 3891/3892: The Session Initiation Protocol (SIP)

'Replaces' Header/ Referred-By Mechanism

RFC 3398: Integrated Services Digital Network (ISDN)

User Part (ISUP) to SIP Mapping

RFC 3966: The telephone URI for telephone numbers (url

tel not supported)

RFC 4497: Inter-working between SIP and QSIG

RFC 5373: Requesting Answering Modes for the Session

Initiation Protocol

RFC 4244: An Extension to the Session Initiation

Protocol (SIP)for Request History Information

RFC 3326: The Reason Header Field for the Session

Initiation Protocol (SIP)

RFC 3428: Session Initiation Protocol (SIP) Extension

 for Instant Messaging (partial)

RFC 3608: Service Route header

RFC 3327: Path Header

RFC 2246: The TLS Protocol Version 1.0

RFC 3268: Advanced Encryption Standard (AES) Cipher

suites for Transport Layer Security (TLS)

RFC 3280/5280:

Internet

Revocation List (CRL) Profile

X.509

Public

Key Infrastructure

Certificate

and

Certificate

RFC 3711: The Secure Real-time Transport Protocol (SRTP)

(media integrity)

RFC 4568: Session Description Protocol (SDP) Security

Descriptions for Media Streams

RFC 5806: Diversion Indication in SIP

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

7.1.2

RTP, T38 & DTMF (used for SIP)

RFC 2617: HTTP Authentication : Basic and Digest Access

Authentication

RFC 1321: Authentication for Outgoing calls

RFC 2833/4733: DTMF Transparency. RFC 2833 replaced by

RFC 4733

RFC 3842: A message Summary and Message Waiting

Indication Event Package

RFC 4028: The session timers in the Session Initiation

Protocol

RFC 3725: Best current practices for Third party Call

Control (3 pcc) in SIP (scenario 1). Invite

without SDP.

RFC 3960: Early Media (partial): Gateway model not

supported

RFC 1889/1890: RTP : A transport protocol for Real-Time

applications

RFC 2198: RTP Payload for Redundant Audio data

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RFC 3550: RTP: A Transport Protocol for Real-Time
application (audio only)

RFC 3551: RTP Profile for Audio and Video Conferences

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with Minimal Control (audio only)

RFC 3711: The Secure Real Time. Supported on A-LU IP

Phone and Softphone

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RFC 3362: T38 ITU-T Procedures for real time Group3 Fax
Relay / communications over IP

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7.2

SIPMOTOR processes

In the OmniPCX Enterprise, the logical functions of

registrar, location service, proxy server and gateway
are

co-located in the process called sipmotor, running on

the CPU7/CS2/AS board.

You may use the linux ps command to verify that the SIP
processes are running :

Example in R9.1:

(1)OXE> ps -edf

root

2247

root

2248

root

2249

root

2250

root

2251

| grep sip

820 0 Jan05

2247 0 Jan05

2247 0 Jan05

2247 0 Jan05

2247 0 Jan05

00:00:00

00:00:31

00:00:00

00:00:00

00:00:00

/DHS3bin/servers/sipmotor

/DHS3bin/servers/sipmotor

/DHS3bin/servers/sipmotor

/DHS3bin/servers/sipmotor

/DHS3bin/servers/sipmotor

00:00:00

00:00:00

00:00:00

00:00:00

00:00:00

[#sipmotor]

[sipmotor_tcl]

[sipmotor]

[sipmotor_dump]

[sipmotor_presen]

Example in R10.0:

(1)OXE> ps -edf

root

2202

root

2203

root

2204

root

2205

root

2206

| grep sip

801 0 2011

2202 0 2011

2202 0 2011

2202 0 2011

2202 0 2011

All processes can be forced to reset with the command:

dhs3_init -R SIPMOTOR, this command stops properly the

SIPMOTOR processes and restarts

them.

(1)OXE> dhs3_init –R SIPMOTOR

They will be automatically relaunched after a few

seconds.

The next commands can be used as well:

killall sipmotor, this command kills the SIPMOTOR

processes and restarts them.

kill -9 “father pid”, this command kills the SIPMOTOR

processes and restarts them.

If no licenses about SIP are present, the SIPMOTOR

processes are not running.

7.3

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OXE duplication

In case of OXE duplication, the SIPMOTOR is complety

started on the Stand-By CPU, but acting as StandBy
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(cannot treat the SIP requests). The Main CPU puts up to
date the Stand-By CPU about the SIP contexts

(Calls, registrations, subscriptions, etc...). In case

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of bascul, the SIP phone calls are maintained and the

registration and subscriptions are kept.

In Case of spatial redundancy with dual subnetworks (2

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main IP addresses), the SIP is using the FQDN of

the OXE (nodename + DNS local domain name) for the SIP
messages and also for the responses of the SIP

messages, in that case, the remote SIP equipment must be

use it. A use of external DNS server is

recommended to resolve this FQDN.

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Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

7.4

The OXE contains the following compoments:

7.4.1

Registrar

Register the addresses of the SIP terminals (“Location

Service”)

The REGISTRAR is containing in the file “localize.sip”

under /tmpd. If for any reasons you need to

clear all entries in the registrar database, remove this

file and then restart the SIPMOTOR:

(1)OXE> rm /tmpd/localize.sip

(1)OXE> dhs3_init -R SIPMOTOR

7.4.2

Proxy

Entity between the Client and the Server, the proxy is

used to route the SIP requests.

The call can be routed between 2 SIP terminals, for

instance Alice calls Bob (both are SIP), in that

case, Alice sends a SIP request to the proxy, and the

proxy sends this request to Bob.

The proxy can be used only for the authentication of the

SIP equipment for Registration or SIP

request.

The proxy can modify the request by adding information

like a Via, Record-route, etc...

INVITE with leg1

INIVTE with leg1

Alice

Proxy

UAC

Bob

UAS

The INVITE is the same on each proxy sides, to get this

behavior, and the UAC manage the IP address of

the OXE SIP proxy as the “Outbound proxy”

Here an example:

The UAC IP address: 172.27.143.184

 The proxy IP address: 172.27.143.186

The UAS FQDN: oxe-ov.alcatel.fr (IP address:

172.27.141.151)

Fri Jun 29 14:08:10 2012 RECEIVE MESSAGE FROM NETWORK

(172.27.143.184:5060 [UDP])

----------------------utf8----------------------INVITE
sip:172.27.143.186 SIP/2.0

Via: SIP/2.0/UDP
172.27.143.184:5060;rport;branch=z9hG4bKPjX7-
GJh79mg04nEbZ0yxYsWP3MCiy4C4H

Max-Forwards: 70

From: <sip:[email protected]>;tag=BJ2er-
g.ONc2M.MQJ9qO.wfpLyp8qfQ3

To: <sip:[email protected]>

Contact: <sip:[email protected]:5060>

Call-ID: L9TrfBGqqYwgo6CR.c9YtaiyulB9OGVU

CSeq: 23308 INVITE

Route: <sip:oxe-ov.alcatel.fr;transport=udp;lr>

Route: <sip:[email protected];transport=udp>

Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE,

SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

Supported: 100rel, norefersub

User-Agent: OmniTouch 1.5.13.7

Content-Type: application/sdp

Content-Length:

283

The OXE SIP proxy receives an INIVTE with the

information “Route” corresponding to the final end point
for

the SIP call. In that case, the OXE SIP proxy acts like
a proxy (not a back to back). Due to this, the proxy

sends the next INVITE to the final SIP endpoint.

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Fri Jun 29 14:08:10 2012 SEND MESSAGE TO NETWORK

(172.27.141.151:5060 [UDP]) (BUFF LEN = 1130)

----------------------utf8----------------------INVITE
sip:[email protected];transport=udp SIP/2.0

Route: <sip:oxe-ov.alcatel.fr;transport=udp;lr>

Record-Route: <sip:172.27.143.186;lr;transport=UDP>

Via: SIP/2.0/UDP
(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
172.27.143.186;branch=z9hG4bK1053e27e7fdda06c573798bc91cd12

Via: SIP/2.0/UDP
(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
172.27.143.184:5060;received=172.27.143.184;rport=5060;bran

Max-Forwards: 69

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
From: <sip:[email protected]>;tag=BJ2er-
g.ONc2M.MQJ9qO.wfpLyp8qfQ3

To: <sip:[email protected]>

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Contact: <sip:[email protected]:5060>

Call-ID: L9TrfBGqqYwgo6CR.c9YtaiyulB9OGVU

CSeq: 23308 INVITE

Allow:
PRACK,INVITE,ACK,BYE,CANCEL,UPDATE,SUBSCRIBE,NOTIFY,REFER,M

Supported: 100rel,norefersub

User-Agent: OmniTouch 1.5.13.7

Content-Type: application/sdp

Content-Length: 283

Session-Expires: 1800

The proxy is adding some information on the INVITE sent

to the final SIP end point, but the INVITE is the

same than the one received (same Call-ID, same FROM,

same TO, same TAGs, etc...)

The REQUEST-URI has been modified according to the

information from the Route from

the first INVITE.

INVITE sip:[email protected]

Information added:

Via: SIP/2.0/UDP 172.27.143.186;

branch=z9hG4bK1053e27e7fd…

Correponding to the proxy “identification“

Record-Route: <sip:172.27.143.186;lr;transport=UDP>

Correponding to the path for the answers (the answers

must be sent to this

IP address)

Session-Expires: 1800

Corresponding to the session timer used on the proxy

The Proxy can be used as a Back-to-Back, in that case,

on each side, two different legs will be found

INVITE with leg2

INIVTE with leg1

Alice

UAC

Proxy

UAS

Bob

UAC

UAS

Two different INVITEs on each proxy sides.

There are no specific information on the INVITE, because

the proxy is acting as an UAS for the caller

and an UAC for the called party.

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

7.4.3

Gateway

Entity between SIP world to legacy world, the gateway is

used to establish a call from a SIP equipment to an

ISDN link, to a legacy set, etc… and vice versa

Do not confuse the SIP gateway with the OmniPCX

Enterprise media gateway boards. The SIP

gateway is a logical entity that resides within the call

server (CS) and is responsible for the SIP

signaling for the conversation setup, while the media

gateway boards (GD, GA, INTIP) are the

physical devices where the media session will be

established when calling to a classic PBX set.

There is one and only one internal SIP gateway. But

there can be many different external SIP

gateways (we will come back to this in a later section).

The SIP gateway is associated to a SIP trunk group.

Although there can be many SIP Trunk Groups,

there is only one SIP trunk group which is associated to

the local SIP gateway. We call this special

trunk group the local SIP trunk group.

7.4.4

Dictionnary

Contains the SIP users created on the OXE, it is the

database that holds the mapping between SIP URLs

and PBX directory numbers (MCDUs). Each registered SIP

terminal is automatically added to the

dictionnary. Classic PBX terminals are added only if a

SIP URL is defined for them in the user management.

Most of the time you shouldn‟t do anything with the

Dictionnary. Everything will be handled

automatically. You need to access the SIP Dictionnary

configuration only for configuration of aliases.

7.4.5

SIP users

On the OXE , we have the two types of SIP users:

SIP Device

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The SIP device is considered as an external SIP user, it
means that the SIP device is linked

to the local SIP gateway, and use its configuration

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The phone features are limited

SIP Extension(or SEPLOS)

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o

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The SIP extension is considered as an internal SIP user,
it means that the SIP extension

can be access to some OmniPCX Enterprise services and

phone features

It can used some OmniPCX Enterprise‟s prefixes, can be

declared as a room set, etc…

The phone features available depend also of the SIP

phone itself.

A SIP extension is attached to a virtual UA board, idem

as an IPtouch.

On OXE, it is necessary to understand that a SIP

extension user is different than the SIP phone
associated

to this user, for instance:

- If the SIP phone is fowarded, that doesn‟t mean that

the user is forwarded.

- If the user is forwarded, that doesn‟t mean that the

SIP phone is forwarded.

It is very important to reminder this behaviour.

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The declaration of a SIP user binds the information

configured in the SIP set with the information stored
into

the database of the OmniPCX Enterprise.

If you don‟t fill in the SIP part in the OmniPCX

Enterprise user configuration, the default values will
be :

URL User Name = MCDU of the user.

URL Domain = SIP domain name of the OmniPCX Enterprise,

i.e. the SIP set is considered as

registered on the OmniPCX Enterprise.

This is usually exactly what we want so you shouldn‟t

modify anything here.

After the creation of the user a corresponding entry

will automatically be added to the SIP Dictionnary.

Note: The value for the URL (<username>@<domainname>)

configured on the SIP set and in the OmniPCX

Enterprise SIP Dictionnary MUST match. This can be an

issue if you modified one of these parameters by

hand and not the other one.

7.4.6 SIP External Voice Mail

On the OXE, it is possible to connect external voice

mail, as the OmniTouch 8440, to be able to manage it

and use it, the local SIP gateway must be managed first.

7.5

 Overview of Interactions between Components

The following diagram shows the relations between the

functional SIP modules in OmniPCX Enterprise :

Dictionnary

Registrar

sip : [email protected]

is reachable at

phone2.alcatel-lucent.com

sip : [email protected]

is reachable at

phone1.alcatel-lucent.com

Gateway

Proxy

Legacy set

sip : [email protected]

phone1.alcatel-lucent.com

Ed. 07

18

sip : [email protected]

phone2.alcatel-lucent.com

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

7.6

Network number rules

The OXE is using network (or subnetwork or routing

tables) for different applications and must be unique
for

each application. It is very important for SIP to

respect the next configuration:

The ABC-F network is using its one network number

(managed on System parameter).

The VPN are using different network numbers according to

the configuration.

The local Hybrid Link (for CCD) is using its one network
number.

For the local SIP gateway, it is necessary to use a

network number used only for it, do not use a

network number used by another application.

Each external ABC-F gateways are using their one network

number.

These rules must be respected to avoid SIP issues.

7.7

SIP parameters explanation / under the object SIP:

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
7.7.1

SIP Trunk Group

(idem for R9.1 and R10.x)

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The SIP Trunk Group is mandatory if you want to use the
Local SIP gateway or an external SIP gateway (not

necessary for SEPLOS users).

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The Trunk Group is used to give channels for SIP calls,
according to its type and configuration, the features

available are differents.

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Different types of SIP trunk Groups are available on
OXE:

The SIP ABCF Trunk Group.

Maximum of features available, the number of accesses is

from 2 to 32 (31 channels

for one access).

The SIP ISDN Trunk Group.

Less features available compares to ABCF Trunk Group,

the number of accesses is

from 2 to 32 (31 channels for one access).

The Mini SIP ABCF Trunk Group.

Maximum of features available, the number of accesses is

from 2 to 32 (2 channels

for one access).

The Mini SIP ISDN Trunk Group.

Less features available compares to ABCF Trunk Group,

the number of accesses is

from 2 to 32 (2 channels for one access).

Level of service depending on used trunk group :

o Call transfer

ISDN

:Using re-INVITE in the opened dialog.

ABC-F :Via REFER, « referred-by » and « replaces ».

o Call forward

ISDN

:Done internally.

ABC-F :Redirecting with 3xx. New call has to be

performed by remote party.

o Call discrimination

ISDN

:Same as ISDN.

ABC-F :No discrimination.

Ed. 07

19

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

To create an SIP Trunk Group, go under /Trunk Groups

Trunk Group Type

: Select T2 for all the different types of SIP Trunk

Group

Trunk Group Name

: Manage a name for the SIP Trunk Group

Number Compatible With

: Keep “-1” everytime, don‟t manage another value

 Remote Network

: Enter a Remote network number, for an ABCF TG, use the

dedicated number, for ISDN TG

keep 255 (idem as legacy T2 ISDN Trunk group)

Node number

: Enter the node number of your OXE

Q931 Signal variant

: - For an ABCF SIP Trunk group, select ABC-F

- For an ISDN SIP Trunk Group, select ISDN

Number Of Digits To Send

: Keep “0” everytime, don‟t manage another value

T2 Specification

: - Select SIP for a SIP Trunk Group (ISDN or ABCF)

- Select Mini SIP for a Mini SIP Trunk group (ISDN or

ABCF)

Public Network COS

: According to the value manage, the OXE will use the

rights of the associated category

DID transcoding

Group)

: This parameter is set to “True” only in case of ISDN

SIP Trunk Group (or Mini SIP ISDN Trunk

Associated Ext SIP gateway

: Enter the external SIP gateway used if there is no DCT

managed on the ARS route, the DCT

from the ARS route is used in priority (From R10.1)

To create a SIP Trunk Group, go under /Trunk

Groups/Trunk Group

IP Compression Type

: - “Default” means only the system algorithm used on

SDP

- “G711” means the use of the sytem algorithm and the

PCM with the system law

Trunk COS

: According to the value manage, the OXE will use the

rights of the associated category

IE External Forward

: Select “Diverting leg information” if you want to use

the History-Info or Diversion header (only

from R10.x for Diversion header)

To create an SIP Trunk Group, go under /Trunk

Groups/Trunk Group/Virtual accesses for SIP

Number of SIP Accesses

: Enter the number of SIP accesses needed on the SIP TG

(value from 2 to 32)

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Some other parameters can be modified with Alcatel-
lucent's agreement according to the

AAPP

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tests(applications and phones) and/or the SIP
Interoperability tests (SIP providers).

Ed. 07

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20

TG0069

OmniPCX Enterprise

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Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

7.7.2

The local SIP gateway

(idem for R9.1 and R10.x)

Used for the local SIP users (SIP Device) and the
external Voice mail

To manage the Local SIP gateway, go under /SIP/SIP

Gateway

SIP Subnetwork

: Corresponds to the local SIP network (different than

the ABC-F network and used only for the

local SIP gateway).

SIP Trunk Group

: Corresponds to the SIP Trunk group (better to use an

ABCF SIP Trunk group)

IP Address

: Corresponds to the IP address of the CPU (autofill)

Machine name – Host

: Corresponds to the nodename associated to the main IP

address (managed via netadmin autofill).

SIP Proxy Port Number

: Corresponds to the SIP port number (by default 5060).

SIP Subscribe Min Duration

: Corresponds to the minimum duration of a SIP

subscription (for message waiting indication or

for result of a transfer).

SIP Subscribe Max Duration

: Corresponds to the maximum duration of a SIP

subscription (for message waiting indication or

for result of a transfer).

Session Timer

: Corresponds to the timer value to supervise an active

SIP session. A RE-INVITE or UPDATE

message is sent before SIP Session Timer expiry (for all

SIP elements).

Min Session Timer

: Corresponds to the mimimum session timer value

accepted by the OXE. When a SIP call is

established, the session timer is negociated between the

two parties.

Session Timer Method

: Corresponds to the method used for session timer, the

OXE sends a RE-INVITE or an

UPDATE message.

DNS local domain name

: Corresponds to local DNS suffix used for SIP. The FQDN

of the OXE is the nodename + this

domaine name (mandatory in case of spatial redondancy).

DNS type

 : Corresponds to the DNS mode (A or SRV).

SIP DNS1 IP Address

: IP address of the first DNS server. (Not manage the

CPU IP address)

SIP DNS2 IP Address

: IP address of the second DNS server. (Not manage the

CPU IP address)

SDP in 18x

: Used to put SDP information on th 18x sent by the OXE.

Cac SIP-SIP

: To allow or not, the domains control in SIP to SIP

communications.

INFO method for remote extension

: Using the INFO method for DTMF in case for the Nokia
Call Connect (NCC) only.

Dynamic Payload type for DTMF

: Payload value used for DTMF, default value 97 (used by

the SIP device for instance).

Ed. 07

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TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

7.7.3

The external SIP gateways

Used to connect external SIP equipments // applications

(SIP provider, Call centre application, etc…).

SIP External Gateway ID

: Id of the gateway

Gateway Name

: Name given to the gateway

SIP Remote domain

: IP address or FQDN of the remote SIP equipment (if

FQDN, need to use a DNS server)

PCS IP Address

: PCS IP address used to backup this gateway in case of

link failure with the CPU

SIP Port Number

: SIP port number used to send SIP messages on the

remote gateway

SIP Transport Type

: Transport type for SIP messages (UDP or TCP)

Belonging Domain

: Used to define the domain part of the URI (FROM and

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
PAI) on the SIP message

Registration ID

: Registration id used on the user part if the remote

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gateway needs it

Registration ID P_Asserted

: Used the registration ID on the P_Asserted Identity

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(PAI)

Registration timer

: Timer used for registration (0 = no registration)

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SIP Outbound Proxy

: Send the messages (INVITE and REGISTER) on this

address

Supervision timer

: Used to supervised the remote gateway (OPTION message

sent)

Trunk group number

: SIP trunk group used for this SIP gateway

Pool Number

: Can associate 2 external SIP gateways in one pool

(Load Balancing)

Outgoing realm

: Realm of the remote gateway (Outgoing messages

authentication)

Outgoing username

: Username from the remote gateway (Outgoing messages

authentication)

Outgoing Password

: Password from the remote gateway (Outgoing messages

authentication)

Incoming username

: Username used by the remote gateway (Incoming messages

authentication)

Incoming Password

: Password used by the remote gateway (Incoming messages

authentication)

RFC 3325 supported by the distant

: PAI supported for Outgoing calls

DNS type

: DNS requests types (A or SRV)

SIP DNS1 IP Address

: IP address of the first DNS server (Not manage the CPU

IP address)

SIP DNS2 IP Address

: IP address of the second DNS server (Not manage the

CPU IP address)

SDP in 18x

: Used to put SDP information on th 18x sent by the OXE

Minimal authentication method

: Used to activate or not the authentication (DIGEST or

SIP none)

INFO method for remote extension

: Using the INFO method for DTMF in case of remote

extension

Send only trunk group algo

: Used to send only the algorithm managed on the SIP TG

To EMS

: Used to activate the RFC4916 (Add specific fields for

identification on EMS)

SRTP

 : Used in case of SIP TLS to select the RTP mode

(secured or not) (From R10.0)

Routing Application

: - False: SDP sets on the SIP messages (INVITE,

200ok...)

- True: No SDP on the SIP messages, this parameter is

used for some specific configuration for

carriers

Ignore inactive/black hole

: Only for SIP ABC-F.

Ed. 07

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

- False means that the receipt of a Re-INVITE, whose SDP

indicates either inactive or c=0.0.0.0

is handled as an Hold request.

- True means that the same kind of Re-INVITE leads the

RTP flow towards the remote party to

be cut.

Contact with IP address

: In case of spatial redundancy with dual subnetworks,

the IP address of the main Call

Server is put on the Contact field instead of the FQDN

of the OXE

Dynamic Payload type for DTMF

: Corresponds to the payload value for DTMF must be the

same than value from the remote SIP

equipment.

100 REL for Outbound Calls

: - Not supported : Outbound INVITE doesn‟t indicate

100Rel parameter.

- Supported : Default Value. Outbound INVITE indicates

100Rel in “Supported” header.

- Required : Outbound INVITE indicates 100Rel in

“Require” header.

100 REL for Incoming Calls

: - Not requested : Default value. 18x response

triggered from OXE doesn‟t indicate 100Rel in

“Require” header.

- Required mode1 : 18x response triggered from OXE

indicates 100Rel in “Require” header

only if it provides SDP.

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- Required mode2 : 18x provisional response triggered
from OXE indicates 100Rel in “Require”

header.

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Gateway type

it is not

: Use to define if the remote SIP gateway is un Open

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Touch or not, keep default configuratiuon if

a Open Touch (From R10.0)

Re-Trans No. for REGISTER/OPTIONS : Number of

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retransmission of SIP REGISTERs/OPTIONs messages, from 1
to 10 (From R10.0)

P-Asserted-ID in Calling Number

: - If True, Calling Number is filled from P-Asserted-ID

header

- If False, Calling Number is filled from FROM header.

(From R10.0)

Trusted P-Asserted-ID header

header)

: Octet3a_Calling is filled based on this parameter

(Used, only when there is P-Asserted-ID

(From R10.0)

Diversion Info to provide via

: In the Outbound INVITE the selected Header is added to

provide information about Call

deflection/forward. The OXE can use History-Info (RFC

4244) or Diversion (RFC 5806)

(From R10.0)

Outbound calls only

: - if False, the existing procedure applies.

- If True, the External Gateway is skipped during the

lookup procedure of the origin of the call.

The way to determine the origin of an inbound call, e.g.

the External Gateway it comes from, is

made in such a way that in that topology, the lowest

External Gateway, in term of numbering, is

chosen.(From R10.1)

SDP relay on Ext. Call Fwd

: In case of SIP trunk to SIP trunk call rerouting

(essentially external to external call forward), in

order to adapt specific SIP profile, OXE offers the

possibility to transit SDP answers received in

180 or 183 on outgoing leg only in 180 answer on

incoming leg.

- Default : normal procedure apply. SDP can transit with

183 message depending on call flow.

- 180 only : any SDP received in 180 and 183 on outgoing

leg will not transit on incoming leg in

183 provisional answer but only in 180 ringing one.(From

R10.1)

Trusted From header

: Octet3a_Calling is filled based on this parameter

(Used, only when there is no P-Asserted-ID

header). To be used when calling number is found in FROM

header and should be considered

as trusted by the system. (From R10.0)

Support Re-invite without SDP

: - if True, the OXE will send a REINVITE without SDP in

case of supervised transfer between

two SIP calls, only if the SIP equipment support it.

 - if False, the OXE will send a REINVITE with SDP. (From

R10.1)

Restriction : When PRACK is supported, this parameter

must be ckeched at False

Proxy identification on IP address

: - if True, a dynamic “DNS cache” per SIP External

Gateway is handled by OXE to store the IP

address(es) to where Register and further INVITE may be

sent. At the beginning of the

procedure, this DNS cache is empty. (From R10.1)

Registration on proxy discovery

: - if True, used when SIP Carrier provide more than one

outbound proxy. As soon as, on carrier

side a switch happen from one proxy to anthoer, calls

cannot be neither delivered to OXE, nor

accepted by the carrier as long as a new registration is

not triggered by OXE. (From R10.1)

Nonce caching activation

Ed. 07

: (From R11)

23

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

RFC 5009 supported / Outbound call

: (From R11)

FAX Procedure Type

: (From R11)

Type of codec negotiation

: (From R11)

7.7.4

Timer usage for SIP Trunking (Trunk Categoy, by default

31)

This only applies to SIP Trunking Call Handling where

generic timers are used

Timer

Timer T302

Timer T303

Timer T304

Timer T305

Timer T308

Timer T309

Timer T310

Timer T313

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Timer T306

Timer T314

Timer T383

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Timer T389

Timer T392

Timer T397

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7.7.5

Value

15s

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10s

90s

4s

4s

90s

20s

4s

6s

2s

5s

8s

1s

5s

Meaning

Related to SETUP_ACK

Related to Call Process

Related to INFO

Related to Disconnect

Related to Release Complete

Related to ALERT

Related to Connect_ACK and Path Replacement

The SIP proxy

Used to activate some parameters linked to the Proxy

(SIP authentication for instance)

SIP initial time-out

: This attribute specifies the initial value in

milliseconds of the request/reply SIP message

retransmission timeout corresponding to T1. Default

value 500ms

SIP timer T2

: This attribute specifies the maximum time in

milliseconds between two SIP message

retransmissions. Default value 4000ms

Dns Timer overflow

: Timer used to overflow from DNS 1 to DNS 2

Timer TLS

: This attribute is used to define the keep alive for

TLS (From R10.0)

Recursive search

: This attribute is used to define the behavior of the

proxy on reception of a redirection message.

(NOT CURRENTLY USED)

- YES: the proxy handles redirection.

- NO: the proxy leaves the caller to handle redirection.

Minimal authentication method

: Activation of the Proxy authentication

- SIP none, there is no authentication

- SIP Digest, the authetication is validated

Authentication realm

: Corresponds to the authentication SIP domain on the

 OXE

Only authenticated incoming calls : Activation of the

SIP authentication for incoming calls

Framework Period

: Indicates the basic time for an observation period

before to put the IP address in quarantine (3s by

default).

Framework Nb Message By Period

: Indicates the maximum number of received messages

during the time of the observation

periods which may put the IP address in quarantine (25

messages by default).

Framework Quarantine Period

Ed. 07

: Indicates the periods number before to put the IP

address in quarantine (1800s by default)

24

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OmniPCX Enterprise

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TROUBLESHOOTING GUIDE No.0069

TCP when long messages

: This parameter is used when UDP is used as transport

protocol, to allow or not the use of TCP for

long messages. This parameter applies to external

gateways, SIP extensions, SIP devices and SIP

external voice mails.

- True (default value): TCP is used, rather than UDP,

when the message size is higher than the

maximum size (1300 bytes)

- False: UDP is used, whatever the size of messages.

Retransmission number for INVITE : This Attribute

corresponds to the number of INVITE retransmission, from
1 to 6 (From R10.0)

SIP timers explanation:

Timer

Timer 1

Timer 2

Value

500 ms

4000 ms

Timer 4

Timer A

Timer B

Timer C

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Timer D

5000 ms

Initially T1

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64 *T1

> 3 min

32s for UDP

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0s for TCP

Initially T1

64 *T1

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Initially T1

64 *T1

T4 for UDP

0 s for TCP

64* T1 for UDP

0 s for TCP

T4 for UDP

0 s for TCP

Timer E

Timer F

Timer G

Timer H

Timer I

Timer J

Timer K

7.7.6

Meaning

Round-trip time (RTT) estimate

The maximum retransmit interval for non-INVITE requests

and INVITE responses

Maximum duration a message will remain in the network

INVITE request retransmit interval, for UDP only

INVITE transaction timeout timer

Proxy INVITE transaction timeout

Wait time for response retransmits

Non-INVITE request retransmit interval, UDP only

Non-INVITE transaction timeout timer

INVITE response retransmit interval

Wait time for ACK receipt

Wait time for ACK retransmits

Wait time for non-INVITE request retransmits

Wait time for response retransmits

SIP Registrar

(idem for R9.1 and R10.x)

Used to manage the registration timers

SIP Min Expiration Date

: Minimum lifetime of a record accepted by the Registrar

(in secondes). Default value 1800.

SIP Max Expiration Date

: Maximum lifetime of a record accepted by the Registrar

(in secondes). Default value 86400.

The minimum value must not be under 420 (7 minutes). The

REGISTER must not be used for

“keep alive” mechanism. 900 (15 minutes) is a minimum

acceptable value.

7.7.7

SIP Dictionnary

(idem for R9.1 and R10.x)

Corresponds to the SIP users created on the OXE, this

dictionnary is fill up automatically when a SIP user is

 created, entries on this dictionnary can be created

manually if needed (Not used), but the purpose of this

object is to be able to modify one entry already created

or to add aliases

Directory Number

: Corresponds to the directory number of Station,

Network number or Vmail number.

Alias No.

: Can create different alias for the same directory

number

SIP URL Username

: User part of the URL. SIP identifies users by their

URLs (Universal Resource Locator), composed of

a user part and a domain part (user@domain).

Ed. 07

25

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TROUBLESHOOTING GUIDE No.0069

SIP URL Domain

: Domain part of the URL. SIP identifies users by their

URLs, composed of a user part and a domain

part (user@domain). If the domain part is omitted on

creation of a set, the domain part of the

installation URL is used (SIP/SIPgateway).

SIP URL Type

: Corresponds to the user type (SIP extension or SIP

Device).

SIP URL Origin

: Corresponds to the origin node.

7.7.8

SIP Authentication

(idem for R9.1 and R10.x)

Used to modify the password of a entry created

automatically (SIP user for instance)

Directory Number

: Directory number of the entry selected (not

modifiable)

SIP Authentication

: SIP login associated to the entry (not modifiable)

SIP Passwd

: Enter a new password if needed

Confirm

: Confirmation of the new password entered

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
7.7.9

Quarantined IP Addresses

(idem for R9.1 and R10.x)

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Used to put the IP addresses of the SIP equipments you
want to put in quarantined manually, SIP messages

from these addresses are dropped silently.

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
7.7.10

Trusted IP Addresses

(idem for R9.1 and R10.x)

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Used to put the IP addresses of the SIP equipments not
affected by the quarantined mechanism. If after

management the communication with this SIP equipments is

still rejected by the OXE, restart the

SIPMOTOR processes.

7.7.11

SIP To CH Error Mapping

(idem for R9.1 and R10.x)

Used to link the error SIP messages to the ISDN Q850

causes, for each error SIP message, you select one

Q850 cause

A default configuration is done, without specific needs,

no modifications have to be made.

Bad request

Unauthorized

Payment required

Forbidden

Not found

Method not allowed

Not acceptable

Proxy authentication required

Request timeout

Conflict

Gone

Length required

Request entity

...

Ed. 07

Request terminates

Not acceptable here

Server internal error

Not implemented

Bad gateway

Service unavailable

Server timeout

Version not supported

Busy everywhere

Decline

Does not exist anywhere

Not accept

Unallocated number

User busy

No user responding

Call rejected

Invalid number format

No circuit

Temporary failure

Bearer cap. not implemented

Incompatible destination

Others

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Ed. 07

27

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

7.7.12

CH To SIP Error Mapping

(idem for R9.1 and R10.x)

Used to link the ISDN Q850 causes to the error SIP

messages, for each Q850 cause, you select error SIP

message.

A default configuration is done, without specific needs,

no modifications have to be done.

Unallocated number

Channel type not implemented

No route to specify transit NW

Req facility not implemented

No route to destination

Only Rest Digi Info Becap Avail

France Specific

Option not implemented

Denmark Specific

Invalid call reference value

Channel unacceptable

Identified channel does not exist

Call awarded - deliv in estab channel

Susp Call Exists But Call Ident

Reserved MLPP

Call Identity in use

Normal call clearing

No call suspended

User busy

Call having req call ID cleared

No user responding

Japan Specific

No answer from user

Incompatible destination

Call rejected

Invalid transit network selection

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Number changed

Invalid message

Nonselected user clearing

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Mandatory info element missing

Destination out of order

Msg type non-exist or not impl

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Invalid number format

Message not compat with call state

Facility rejected

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Info element non-exist or not impl

Response To STATUS INQUIRY

Invalid info element content

Normal unspecified

Recovery on timer expiration

No circuit

Protocol error

Network out of order

Interworking

Temporary failure

...

7.8

SIP parameters explanation / under the object USERS:

7.8.1

Not found

Gone

Temporarily unavailable

Address Incomplete

Busy here

Not acceptable here

Server internal error

Not implemented

Bad gateway

Service unavailable

Decline

Others

SIP Device

The SIP Device is used for voice SIP calls and FAX SIP
calls, the SIP Device is considered as an External

SIP user, so the features are limited (same as SIP TG)

SIP Device creation

Directory Number

: Corresponds to the directory number of the SIP Device

: Select “SIP device” for the type of set

Set Type

URL UserName

: The user name corresponds to the SIP Device directory

number - autofill

URL Domain

: Corresponds to the OXE domaine name (nodename) -

autofill

SIP Authentication

: The user name corresponds to the SIP Device directory

number – autofill

External Gateway Number

: Used in case of Open Touch configuration, determine

the external Gateway number to reach the OT

(From R10.0)

Gateway type

R10.0)

 : Used in case of Open Touch configuration, determine

the gateway type to reach the OT (From

In normal use, only the Directory Number and the set

type are managed, the other parameters can

be modified only if needed

The SIP device is linked to the local SIP gateway

The local SIP gateway must be managed and is in service

to be able to make and receive calls

Ed. 07

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TROUBLESHOOTING GUIDE No.0069

With the current Linux OS, OXE has a limitation in

handling more than 1000 data equipment if it

is connected in the same sub-network. So we need to have

a seperate VLAN in between to

handle this. OXE CS must be placed under separate subnet

and the IP Phones distributed

under different other subnets

All unnecessaries subscriptions must be deactivated on

SIP Devices when service is not

available on OXE, example: Voicemail, notification …

7.8.2

SIP Extension (or SEPLOS)

The SIP Extension is used only for voice calls, is

considered as an Internal SIP user, so it is possible to
use

phone features and facilities from the OXE.

It is not necessary to manage the local SIP gateway if

you want to use it, only the proxy has to be (for

authentication)

SIP Extension creation

Directory Number

: Corresponds to the directory number of the SIP

Extension

: Select “SIP extension” for the type of set

Set Type

URL UserName

: The user name corresponds to the SIP Extension

directory number - autofill

URL Domain

: Corresponds to the OXE domain name (nodename) -

autofill

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
SIP Authentication

: The user name corresponds to the SIP Extension

directory number – autofill

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Other SIP extension parameters

- Under /users/ IP SIP Extension:

Set Type

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: Type of set displayed (SIP extension or SIP device)

IP Address

: IP address of the SIP equipment displayed (information

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
retrevies from the registrar)

- Under /users/ SIP Extension Parameters:

Phone COS

(explanation later)

: Corresponds to the SIP phone class of service and not

the “normal” phone class of service

The SIP extension can be created as a “business” user or

“room” user in case of hospitality. One of the
difference, it that in case of

“business” mode, the SIP extension is multiline (not

manageable) and in case of “room” mode , the SIP
extension is monoline.

7.9

SIP parameters explanation / under the object SIP

Extension:

Used to manage some specific phone features for SIP

extension

Display UTF-8

: Used to display UTF-8 name, if the SIP phone is

compatible,

- if True, the OXE will send the name in UTF-8 to the

SIP Phone

- if False, the OXE will send the “normal” name to the

SIP phone

Display call server information

: Display information on the set display, for instance

if the set is fowarded by using an OXE prefix

- if True, the OXE will send a SIP message MESSAGE

- if False, the OXE will not send this SIP message

The SIP phone must be compatible with the SIP messages

or they will be rejected (405 message).

Ed. 07

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TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Example of a message:

----------------------utf8----------------------MESSAGE
sip:[email protected]:5060;transport=udp;user=phone
SIP/2.0

Supported: replaces,timer,100rel

User-Agent: OmniPCX Enterprise R9.1 i1.605.23

Content-Type: text/plain;charset=UTF-8

To: <sip:31031@myoxe;user=phone>

From: " "

<sip:31031@myoxe;user=phone>;tag=40cc45387a17217352a366b1cf

Call-ID: [email protected]

CSeq: 1185999967 MESSAGE

 Via: SIP/2.0/UDP
172.27.141.151;branch=z9hG4bKb46b58bf397dae02629301df568a1b

Content-Length: 26

Immdiate fwd -> 31000

------------------------------------------------Keep
Alive

: Used to implement the keep alive mechanism between the

OXE to the SIP phone, if the SIP phone

is compatible

- if True, the OXE will send an OPTION message to the

SIP phone

- if False, the OXE will not send this OPTION message

The keep alive timer is managed on the IP Quality Of

Service COS, assoicated to the IP domain of the SIP
Extension user

(seen later)

Send NOTIFY instead of MESSAGE

: Used to send the synamic state of the SEPLOS SIP

message MESSAGE or with a NOTIFY

SIP message

(From R10.0)

7.10 SIP parameter explanation / under the object

External Voice Mail:

Go under /Applications/ External Voice Mail

Voice Mail Dir.No

Sub Type

: Corresponds to the directory number of the External

Voice Mail.

: - Private (default value): The via header is not used

to determine the origin of incoming calls.

- Public: the via header is used to determine the origin

of incoming calls when other headers do not

match.

URL UserName

: Corresponds to the Voice Mail directory number.

URL Domain

: Corresponds to the nodename of the OXE.

PCS IP Address

: Corresponds to the IP address of the PCS to secure

this external SIP Voice Mail.

SIP Authentication

: Correponds to the login used for the authentication to

the external SIP voice mail

SIP Passwd

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
: Correponds to the password used for the authentication
to the external SIP voice mail

Register On Line Number

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
: Directory number used to access the voice mail service
in record mode. This number is dialed

automatically when the 'Rec.' key is pressed on a set.

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Register URL (Username)

: User part of the URL used for access to the voice mail
service in record mode.

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Register URL (Domain)

: Domain part of the URL used for access to the voice

mail service in record mode.

Register Authentication

mode.

: Correponds to the login used to control access to the

external voice mail service in record

Register Password

mode.

Ed. 07

: Correponds to the password used to control access to

the external voice mail service in record

30

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

External Gateway Number

: Used to manage an entity (SIP Device or External Voice

Mail) behind a Proxy. If different from -1, it

is used as an ½ Outbound Proxy: outgoing calls are

routed to it via its RemoteDomain (Gateway Id)

and its Outbound Proxy. Registration (REGISTER) and

supervision (OPTIONS) are still configurable.

Subscription on registration

: Used if the Subscription is done in the same time than

the Registration or in two different messages.

7.11 SIP parameters explanation / under the object

System:

Go under /System/Other System Param./SIP Parameters

Packetization times per codec

: - If True , as many couple of ptime/maxptime

information available for many codecs.

- If False , a single couple of ptime/maxptime

information available for many codecs.

Via Header_ Inbound Calls Routing

Hardwareless for OTBE

: - If False (default value): The via header is not used

to determine the origin of incoming calls.

- If True: the via header is used to determine the

origin of incoming calls when other headers do not

match with the RemoteDomain of an External Gateway.

: NOT CURRENTLY USED (From R10.1)

Local resources

: NOT CURRENTLY USED (From R10.1)

Loose Route with RegID

: The possibility is offered to accept the call if route

only contains a URI with OXE_address

without user part.

- If True, INVITE without RegID in route header is re-

 routed to the destination corresponding to

ReqURI domain part.

- If False, INVITE is accepted. (From R10.1)

Reject unidentified proxy calls

: As an exceptional procedure for inbound calls, if the

origin of the call cannot be determined, either by

looking up the SIP dictionary, or through any other

procedure (call does not comes from a SIP

External Gateway), and if the Source @IP doesn‟t belong

to the trusted @IP list the call is either

delivered to the Call Handling on the Main Gateway, or

rejected with a 403.Forbidden response.

- If it is set to True, such calls are rejected with a

403.Forbidden response.

- If it is set to False, the call is delivered to the

Call Handling on the Main Gateway. (From R10.1)

Transfer : Refer using single step

- If True, new INVITE without Referred-By is provided

- If False, new INVITE with Referred-By is provided

(From R10.1)

Go under /System/Other System Param./System Parameters

SRTP TLS offer answer mode

: - If True: SRTP according to SDP offer/answer model

- If False: SRTP Oxe centralized SRTP mode

(From R10.0)

TLS signaling possible

: - If True: TLS signaling allowed for SIP gateways /

TLS signaling and SRTP allowed for SIP sets

- If False: TLS signaling not possible for SIP gateways

/ TLS signaling and SRTP not possible for

SIP

(From R10.0)

Accept Mu and A laws in SIP

: From the R9.1, the OXE is using only in G711 the

system law for all SIP calls (inbound calls), thanks

to this parameter, the OXE is able to accept the

G711calls using the other law for inbound calls on

external SIP gateways only.

Go under /System/Other System Param./External Signaling

Parameters

NPD for External Forward

: - If -1: redirection information is sent

- If configured with NPD number used by SIP ISDN Trunk:

see the calling name presentation on the

set display of called phone in case of forward

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Calling Name Presentation

Ed. 07

: - If False: Calling Number is not sent

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- If True: display name to external calls is sent

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
8. IP DOMAINS, CODECS AND PCS

8.1

IP domains rules

A SIP equipment can belong to an IP domain, according to

this configuration, it is able to use some

behaviours from its IP domain (see the TC1277 for IP

domain configuration and restrictions)

The first thing to know, it is that a SIP equipment

doesn‟t belong to an IP domain if its IP address is not

managed, it doesn‟t belong in the IP domain 0 as well

(exept for the SIP extension users acting like IPtouch)

in that case if no management is done, the call is

everytime an extra domain call with an Alcatel-Lucent

equipment.

8.2

System law for PCM codec

The system is accepted only the PCM codec of its law. If

the system is using the A law, only PCMA will be

accepted and used, PCMU will be rejected.

Exception: for SIP external gateways, if the OXE

receives an INVITE with only PCMU, the OXE will accept,

but the voice quality is not guaranteed.

The next parameter must be managed:

/System/Other System Param./System Parameters/Accept Mu

and A laws in SIP

False (default): only the system law is accepted

True: the two laws are accepted

8.3

Codecs on SDP

When a SIP call is done, the OXE manage the SDP

according to the next information:

8.3.1

Initial offer : the offer sent in an initial INVITE

The codec list proposed in an initial SDP offer is build

according to the algorithm of the outgoing SIP Trunk

Group.

The outgoing SIP Trunk Group is the one managed in ARS

route or Network/Routing number, NOT the one

managed on the External SIP Gateway.

This codec list is ordered taking into account calling

user extra domain compression law.

Exception : if the caller is a SIP device or a SIP

trunk, the codec list is in the same order than the one

received from the calling party.

SIP trunk algo must be interpreted as « the best

algorithm supported on the trunk » or « the higher

bandwidth consumption supported on the trunk» :

SIP trunk algorithm : default

- The Trunk Group has low capacity. Only G729/G723 is

possible.

 SIP trunk algorithm : G711

Ed. 07

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TROUBLESHOOTING GUIDE No.0069

The Trunk Group supports high bandwidth calls and as a

consequence low bandwidth calls

too. Both G711 and system codec (G729/G723) can be used.

Initial SDP offer content, general case (calling party

is not a SIP device nor a SIP trunk).

Trunk Group compression

type

Default

Default

G711

Intra/Extra IP domain

algorithm

With Compression

Without Compression

With Compression

G711

Without Compression

SDP

System algorithm only (G729 for instance)

System algorithm only (G729 for instance)

System algorithm (G729 for instance) in first position

and PCM (A or MU) in the second position

PCM (A or MU) in first position and system algorithm

(G729 for instance) in the second position

8.3.1

nitial answer : the answer to an initial offer on

incoming call

Pre-requisite :

The SIP equipment must at least propose one codec

supported by OXE in its offer.

OXE Trunk Group used for incoming calls (managed in

External SIP Gateway) must be managed

with algo=G711.

OXE always answers with one codec only :

The one proposed in a by the SIP equipment in case of

mono-codec offer.

The best one in case of multicodec offer, taking into

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account :

- SIP equipment list order (calling party prefered

codec).

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- Called party extra-domain codec.

The answer may be send in 18x and/or 200OK depending on

« SDP in 18x » management.

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OXE initial SDP answer summary (incoming trunk group
algo = G711).

SIP equipment SDP Offer

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G729, G711

G729, G711

G711, G729

G711, G729

G711

G711

Intra/Extra IP domain algorithm

With Compression

Without Compression

With Compression

Without Compression

With Compression

Without Compression

Codec use

G729

G729

G729

G711

G711

G711

For SEPLOS users, the OXE is acting as an IPtouch.

8.4

PCS

The SIP is totally operational on PCS; it is able to

secure all types of SIP elements, but the SIP equipment

connected must be tested to be sure that it will be able

to connect and working on the PCS.

In case of spatial redundancy, the nodename manage on

the PCSs must be the same than

the one managed on the CPUs.

Ed. 07

33

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Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

9. CONTENTS OF A SIP MESSAGES (GENERAL VIEW)

On the SIP messages, we can find different information.

According to the type of message, the information

can change or can be adapted.

For instance, with an INVITE we can have this:

HEADER

BODY

INVITE sip:[email protected]:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP
172.27.142.64:5060;branch=z9hG4bK3047297329

From: "31031"
<sip:[email protected]:5060;user=phone>;tag=c0a80101-
17193256

To: <sip:[email protected]:5060;user=phone>

 Call-ID: [email protected]

CSeq: 1 INVITE

Max-Forwards: 70

Supported: timer, P-Early-Media, replaces

Require: 100rel

Session-Expires: 110

Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE

Contact:
<sip:[email protected]:5060;transport=udp;user=phone>

User-Agent: THOMSON ST2030 hw5 fw2.72 00-1F-9F-16-4F-03

Allow-Events: refer,dialog,message-summary,check-
sync,talk,hold

Content-Type: application/sdp

Content-Length: 203

v=0

o=MxSIP 4219058434975324735 4219058434975324736 IN IP4

172.27.142.64

s=SIP Call

c=IN IP4 172.27.142.64

t=0 0

m=audio 6000 RTP/AVP 8 0 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-15

a=ptime:20

a=mptime:20 20 30 20 a=sendrecv

Between the Header and the Body, you have everytime an

empty line

9.1

The HEADER

The header contains the information to establish a SIP

dialog between the UAC and the UAS.

Here the main information given:

- The Request-URI:

INVITE sip:[email protected]:5060;user=phone SIP/2.0

The initial Request-URI of the message SHOULD be set to

the value of the URI in the To field, except if the

recipient (To field) is forwarded.

Request-URI: forward destination

To: forwarded set

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Ed. 07

34

TG0069

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
- The From:

From: "31001"
<sip:[email protected]:5060;user=phone>;tag=c0a80101-
(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
17193256

The From header field indicates the logical identity of

the initiator of the request.

- The To:

To: <sip:[email protected]:5060;user=phone>

The To header field first and foremost specifies the

desired "logical" recipient of the request.

- The Call-ID:

Call-ID: [email protected]

The Call-ID header field acts as a unique identifier to

group together a series of messages. It MUST be the

same for all requests and responses sent by either UA in

a dialog.

- The CSeq:

CSeq: 1 INVITE

A CSeq header field in a request contains a single

decimal sequence number and the request method. The

CSeq header field serves to order transactions within a

dialog, to provide a means to uniquely identify

transactions, and to differentiate between new requests

and request retransmissions. Two CSeq header

fields are considered equal if the sequence number and

the request method are identical.

- The Max-Forwards:

Max-Forwards: 70

The Max-Forwards header field serves to limit the number

of hops a request can transit on the way to its

destination.

- The Via:

Via: SIP/2.0/UDP
172.27.142.64:5060;branch=z9hG4bK3047297329

The Via header field indicates the transport used for

the transaction and identifies the location where the

response is to be sent.

- The Contact:

Contact:
<sip:[email protected]:5060;transport=udp;user=phone>

The Contact header field provides a SIP URI that can be

used to contact that specific instance of the UA for

subsequent requests. Contact header field MUST be

present and contain exactly one SIP URI in any request

that can result in the establishment of a dialog.

- The Supported and/or Require

Supported: timer, P-Early-Media, replaces

If the UAC supports (requires) extensions to SIP that

can be applied by the server to the response.

Ed. 07

If the UAS receives a supported option tags, it is able

 to use them if needed.

If the UAS receives a required option tags, it must use

them or reject the request

35

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Other information can appear on header according to the

SIP equipment type, to know the meaning of them,

check the SIP RFCs

9.2

The BODY

The body contains the message or information used to

openan RTP connection (codec, IP address, etc…)

v=0

o=MxSIP 4219058434975324735 4219058434975324736 IN IP4

172.27.142.64

s=SIP Call

c=IN IP4 172.27.142.64

t=0 0

m=audio 6000 RTP/AVP 8 0 9 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:9 G722/8000

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-15

a=ptime:20

a=mptime:20 20 20 20 20 a=sendrecv

SDP session description consists of session-level

sections.

Each session-level starts by a letter, corresponding to

an information for RTP channel negociation (in voice

cases)

In that example, we have the next information given:

v= : corresponds to SDP version

o= : corresponds to the originator of the session

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
“MxSIP” = username

“4219058434975324735” = sess-id, forms a globally unique

identifier for the session

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
“4219058434975324736” = sess-version, is a version
number for this session description

(increased in case of SDP modification)

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
“IN” = Internet connection type (thru IP network)

“IP4” = IP V4 is used for IP addressing

“172.27.142.64” = IP address of the SIP equipment (for

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
RTP connection)

s= : corresponds to the session name

c= : corresponds to the connection data

“IN” = Internet connection type (thru IP network)

“IP4” = IP V4 is used for IP addressing

“172.27.142.64” = IP address of the SIP equipment (for

RTP connection)

t= : corresponds to the start and stop times for this

session (t= <start time> <stop time>)

Ed. 07

t= 0 0 means that the “timimg” is not used in that case

This field is mandatory on SDP

36

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

m= : corresponds to the media description

o “audio” = media type (audio, video, text,…)

o “6000” = port number used to sent the media stream

o “RTP/AVP” = transport protocol, in that case, it is

RTP

o 8 0 9 18 101 = payloads (codecs)

a= : corresponds to SDP attributes

“a=rtpmap:8 PCMA/8000” = codec PCMA available on this

SIP equipment

“a=rtpmap:0 PCMU/8000” = codec PCMU available on this

SIP equipment

“a=rtpmap:101 telephone-event/8000” = payload for DTMF

“a=fmtp:18 annexb=no” = no VAD available for this call

(annexb)

“a=ptime:20” = packet time (framing)

“a=mptime:20” = maximum ptime accepted

“a=sendrecv” = direction of the call, in that case both

directions

The SDP is generated according to the SIP equipment,

 each SDP is different for each type of SIP equipment

and type of SIP call.

Ed. 07

37

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

10. EXAMPLES OF COMMON SIP FLOWS

10.1

Registration

In an OmniPCX Enterprise context, the call server (CS)

takes the role of the SIP registrar. Registration is

necessary to bind a given SIP URL to a physical address.

External SIP sets register on the registrar with a

SIP REGISTER request.

Note that there may be a short delay of several seconds

between the time the REGISTER message is

received and the time the registrar database is updated.

Without authentication:

31026

. . . . .

OXE

(SIP set)

(Registrar)

IP=172.27.141.210 FQDN=oxe-ov.alcatel.fr

(1) REGISTER

|------------------->|

(2) 200 OK

|<-------------------|

----------------------utf8----------------------REGISTER
sip:oxe-ov.alcatel.fr SIP/2.0

Via: SIP/2.0/UDP 172.27.141.210:22362;branch=z9hG4bK-

d87543-826b1a28d80c8c6b-1--d87543-;rport

Max-Forwards: 70

Contact:
<sip:[email protected]:22362;rinstance=70dae25b3c1e2541>

To: "31026"<sip:[email protected]>

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
From: "31026"<sip:[email protected]>;tag=e2704074

Call-ID: MDMxYzVkZjBiNWY0NTlmODAwMTk2MTdkNzczZjkwOTM.

CSeq: 1 REGISTER

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Expires: 3600

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,

MESSAGE, SUBSCRIBE, INFO

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
User-Agent: SIP Phone

Content-Length: 0

-------------------------------------------------

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
o

The To header field contains the address of record (SIP

URI) whose registration is to be

created. In the example “oxe-ov.alcatel.fr” is the

domain (OXE main IP address or FQDN)

and 31026 the user name.

The Contact header field contains the physical address

(IP address and port) of the record

whose registration is to be created. In the example it

is 172.27.141.210:22362. Note that

if port number would not have been specified it would

have been taken as 5060 by default.

If any other port number than 5060 is used, it must have

to be specified (here 22362).

The Expires field corresponds to the maximum time of

registration on the REGISTRAR, the

SIP equipment msut send a new REGISTER message to stay

on, if not, it will be removed

from it.

The registrar answers with a 200 OK response upon

successful registration.

----------------------utf8----------------------SIP/2.0
200 OK

Contact:
<sip:[email protected]:22362;rinstance=70dae25b3c1e2541>

To: "31026" <sip:31026@oxe-

ov.alcatel.fr>;tag=aacdbedd6291976d33d249b9d1d52820

From: "31026" <sip:[email protected]>;tag=e2704074

Call-ID: MDMxYzVkZjBiNWY0NTlmODAwMTk2MTdkNzczZjkwOTM.

CSeq: 1 REGISTER

Via: SIP/2.0/UDP

172.27.141.210:22362;received=172.27.141.210;branch=z9hG4bK
d87543-e14134135a40db7d-1--d87543Ed. 07

38

TG0069

;rport=22362

Content-Length: 0

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

With authentication:

31026

. . . . .

OXE

(SIP set)

(Registrar)

IP=172.27.141.210 FQDN=oxe-ov.alcatel.fr

|(1) REGISTER

|-------------------->|

|(2) 401 Unauthorized |

|<--------------------|

|(3) REGISTER

|-------------------->|

|(4) 200 OK

|<--------------------|

The first REGISTER is send without the authenication

parameters and the OXE sends a 401 Unauthorized

message to ask the SIP equipment for the authentication

parameters

----------------------utf8----------------------SIP/2.0
401 Unauthorized

WWW-Authenticate: Digest
qop="auth",nonce="a4c9e550459f63fd80764dc69609c482",realm="
ov"

To: "31026" <sip:31026@oxe-

ov.alcatel.fr>;tag=da389f6e785d72b8910a0f2310d68fcc

From: "31026" <sip:[email protected]>;tag=e2704074

Call-ID: MDMxYzVkZjBiNWY0NTlmODAwMTk2MTdkNzczZjkwOTM.

CSeq: 1 REGISTER

Via: SIP/2.0/UDP
172.27.141.210:22362;received=172.27.141.210;branch=z9hG4bK
d87543-826b1a28d80c8c6b-1--d87543;rport=22362

Content-Length: 0

-------------------------------------------------

The WWW-Authenticate field corresponds to the OXE

information about authentication:The

information “Digest” corresponds to the challenge type

The information “qop” corresponds to the "quality of

protection" values supported by the

server. The value "auth" indicates authentication.

The information “nonce” corresponds to control the

integrity of the authentication information

received by the SIP equipment.

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
The information “realm” corresponds to the SIP
authentication domain, only one can be

managed on the OXE.

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
The Register with the authentication information

----------------------utf8----------------------REGISTER
sip:oxe-ov.alcatel.fr SIP/2.0

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Via: SIP/2.0/UDP 172.27.141.210:22362;branch=z9hG4bK-
d87543-e14134135a40db7d-1--d87543-;rport

Max-Forwards: 70

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Contact:
<sip:[email protected]:22362;rinstance=70dae25b3c1e2541>

To: "31026"<sip:[email protected]>

From: "31026"<sip:[email protected]>;tag=e2704074

Call-ID: MDMxYzVkZjBiNWY0NTlmODAwMTk2MTdkNzczZjkwOTM.

CSeq: 2 REGISTER

Expires: 3600

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,

MESSAGE, SUBSCRIBE, INFO

User-Agent: SIP Phone

Authorization: Digest username="31026",realm="oxe-

ov",nonce="a4c9e550459f63fd80764dc69609c482",uri="sip:oxeov

8806dc1b4321b19",cnonce="e53a2b8923348db7",nc=00000001,qop=

Content-Length: 0

------------------------------------------------Ed. 07

39

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

When the registration timer is too brief

31026

. . . . . . . . . .

OXE

(SIP set)

(Registrar)

IP=172.27.141.210

FQDN=oxe-ov.alcatel.fr

|(1) REGISTER

|------------------------------>|

|(2) 423 Registration Too Brief |

|<------------------------------|

|(3) REGISTER

|------------------------------>|

|(4) 200 OK

|<------------------------------|

When the “expires” is too small compares to the OXE one,

the OXE returns the message “423 Registration

Too Brief”, with its timer, in that case, the SIP

equipment sends a new REGISTER with the timer received.

----------------------utf8----------------------REGISTER
sip:oxe-ov.alcatel.fr SIP/2.0

 Via: SIP/2.0/UDP 172.27.141.210:22362;branch=z9hG4bK-

d87543-826b1a28d80c8c6b-1--d87543-;rport

Max-Forwards: 70

Contact:
<sip:[email protected]:22362;rinstance=70dae25b3c1e2541>

To: "31026"<sip:[email protected]>

From: "31026"<sip:[email protected]>;tag=e2704074

Call-ID: MDMxYzVkZjBiNWY0NTlmODAwMTk2MTdkNzczZjkwOTM.

CSeq: 1 REGISTER

Expires: 60

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,

MESSAGE, SUBSCRIBE, INFO

User-Agent: SIP Phone

Content-Length: 0

-------------------------------------------------

The “Expires” value is equal to 60 in that case, and the

minimum value managed on the

OXE is 1800

----------------------utf8----------------------SIP/2.0
423 Registration Too Brief

Min-Expires: 1800

To: "31026"<sip:31026@oxe-
ov.alcatel.fr>;tag=85d8c7828811c12691305052d6ef7f9a

From: "31026"<sip:[email protected]>;tag=e2704074

Call-ID: MDMxYzVkZjBiNWY0NTlmODAwMTk2MTdkNzczZjkwOTM.

CSeq: 1 REGISTER

Via: SIP/2.0/UDP 172.27.141.210:22362;branch=z9hG4bK-

d87543-826b1a28d80c8c6b-1--d87543-;rport

Content-Length: 0

-------------------------------------------------

The information “Min-Expires” correponds to the minimun

registration timer value of the

OXE (manage on the REGISTRAR object)

----------------------utf8----------------------REGISTER
sip:oxe-ov.alcatel.fr SIP/2.0

Via: SIP/2.0/UDP 172.27.141.210:22362;branch=z9hG4bK-

d87543-826b1a28d80c8c6b-1--d87543-;rport

Max-Forwards: 70

Contact:
<sip:[email protected]:22362;rinstance=70dae25b3c1e2541>

To: "31026"<sip:[email protected]>

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
From: "31026"<sip:[email protected]>;tag=e2704074

Call-ID: MDMxYzVkZjBiNWY0NTlmODAwMTk2MTdkNzczZjkwOTM.

CSeq: 1 REGISTER

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Expires: 1800

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,

MESSAGE, SUBSCRIBE, INFO

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
User-Agent:

Ed. 07 SIP Phone

40

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Content-Length: 0

-------------------------------------------------

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

10.2

Session Iniation Protcol (SIP)

The new REGISTER received on the OXE has the value 1800
(the one from the message

423)

De-registration

31026

. . . . .

OXE

(SIP set)

(Registrar)

IP=172.27.141.210 FQDN=oxe-ov.alcatel.fr

(1) REGISTER

|------------------->|

(2) 200 OK

|<-------------------|

When a SIP equipment is stopped, before it has to send a

REGISTER message to be removed from the

OXE REGISTRAR, for this, it has to send a REGISTER with

an “Expires = 0”

----------------------utf8----------------------REGISTER
sip:oxe-ov.alcatel.fr SIP/2.0

Via: SIP/2.0/UDP 172.27.141.210:22362;branch=z9hG4bK-

d87543-826b1a28d80c8c6b-1--d87543-;rport

Max-Forwards: 70

Contact:
<sip:[email protected]:22362;rinstance=70dae25b3c1e2541>

To: "31026"<sip:[email protected]>

From: "31026"<sip:[email protected]>;tag=e2704074

Call-ID: MDMxYzVkZjBiNWY0NTlmODAwMTk2MTdkNzczZjkwOTM.

CSeq: 1 REGISTER

Expires: 0

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,

MESSAGE, SUBSCRIBE, INFO

User-Agent: SIP Phone

Content-Length: 0

-------------------------------------------------

On the REGISTER, we have the “Expires = 0” and the

“Contact”, this contact is used by the

REGISTRAR to know which physical IP address to remove

for this URI (in case of forking).

If the “Contact” is received with a “*”, the REGISTRAR

must removed all the “Contact”

associated.

In case of duplication, when the Main CPU receives a

REGISTER, the SIPMOTOR sends this REGISTER to

the Stand-BY CPU with the next message:

----------------------utf8----------------------REGISTER
sip:oxe-ov.alcatel.fr SIP/2.0

Via: SIP/2.0/UDP
172.27.141.210:22362;received=172.27.141.210;branch=z9hG4bK
d87543-e14134135a40db7d-1--d87543;rport=22362

Max-Forwards: 70

Contact:
<sip:[email protected]:22362;rinstance=70dae25b3c1e2541>

To: "31026" <sip:[email protected]>

From: "31026" <sip:[email protected]>;tag=e2704074

Call-ID: MDMxYzVkZjBiNWY0NTlmODAwMTk2MTdkNzczZjkwOTM.

CSeq: 2 REGISTER

Expires: 3600

Allow: INVITE

Allow: ACK

Allow: CANCEL

Allow: OPTIONS

Allow: BYE

Allow: REFER

Allow: NOTIFY

Allow: MESSAGE

Allow: SUBSCRIBE

Allow: INFO

Content-Length: 0

User-Agent:

Ed. 07 Alcatel-main Registrar

41

TG0069

-------------------------------------------------

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
10.3

Simple call establishement

The following diagram shows the messages sent from a SIP

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
equipment to an OXE user (Not a SIP one)

UAC

UAS

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
31026

OXE

31004

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
(caller). . . . . . . (proxy). . . . . . . . . . . . . .
(callee)

IP=172.27.141.210

FQDN=oxe-ov.alcatel.fr

INVITE

|-------------------->|

100 Trying

|<--------------------|

| Process to contact the callee

|<------------------------------->|

180 Ringing

|<--------------------|

200 OK

|<--------------------|

ACK

|-------------------->|

Media Session

|
<=====================================================>|

BYE

|-------------------->|

200 OK

|<--------------------|

1) The SIP equipment sends an INVITE to the OXE

Mon Jun 25 11:10:17 2012 RECEIVE MESSAGE FROM NETWORK

(172.27.141.210:63016 [UDP])

----------------------utf8----------------------INVITE
sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 172.27.141.210:63016;branch=z9hG4bK-

d87543-4c3f8f26d532b437-1--d87543-;rport

Max-Forwards: 70

Contact: <sip:[email protected]:63016>

To: "31004"<sip:[email protected]>

From: "31026"<sip:[email protected]>;tag=e9708b0f

Call-ID: MzI0MjQ4MmQ5NjMzZTVmZTlmYTE5NTVhMGNiZWI0ODQ.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,

MESSAGE, SUBSCRIBE, INFO

Content-Type: application/sdp

User-Agent: SIP Phone

Content-Length: 417

v=0

o=- 6 2 IN IP4 172.27.141.210

s= SIP Phone

c=IN IP4 172.27.141.210

t=0 0

m=audio 52694 RTP/AVP 18 101

a=fmtp:18 annexb=no

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

a=sendrecv

Ed. 07

-------------------------------------------------

42

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
The INVITE can contain SDP or not. If there is no SDP,
the ACK (after the 200ok) sent must

contain the SDP information

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2) The SIP equipment receives a provisional answer from
the OXE (100 Trying)

Mon Jun 25 11:10:17 2012 SEND MESSAGE TO NETWORK

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(172.27.141.210:63016 [UDP]) (BUFF LEN = 338)

----------------------utf8----------------------SIP/2.0
100 Trying

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To: "31004" <sip:[email protected]>

From: "31026" <sip:[email protected]>;tag=e9708b0f

Call-ID: MzI0MjQ4MmQ5NjMzZTVmZTlmYTE5NTVhMGNiZWI0ODQ.

CSeq: 1 INVITE

Via: SIP/2.0/UDP
172.27.141.210:63016;received=172.27.141.210;branch=z9hG4bK
d87543-4c3f8f26d532b437-1--d87543;rport=63016

Content-Length: 0

-------------------------------------------------

The 100 Trying is a provisional message sent by the OXE,

this message is generated by the

SIPMOTOR directly, it can be concidered as an automatic

answer of an INVITE to avoid

retransmission from UAC.

3) The SIP equipment receives a provisional answer from

the OXE (180 Ringing or 183 Session Progress)

Mon Jun 25 11:10:18 2012 SEND MESSAGE TO NETWORK

(172.27.141.210:63016 [UDP]) (BUFF LEN = 815)

----------------------utf8----------------------SIP/2.0
180 Ringing

Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER,

SUBSCRIBE, OPTIONS, UPDATE

Contact: sip:oxe-ov.alcatel.fr

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

Content-Type: application/sdp

To: "31004" <sip:31004@oxe-

ov.alcatel.fr>;tag=bb28096d41c595340f577a538bf30d54

From: "31026" <sip:[email protected]>;tag=e9708b0f

Call-ID: MzI0MjQ4MmQ5NjMzZTVmZTlmYTE5NTVhMGNiZWI0ODQ.

CSeq: 1 INVITE

Via: SIP/2.0/UDP
172.27.141.210:63016;received=172.27.141.210;branch=z9hG4bK
d87543-88163a3aa534591a-1--d87543;rport=63016

Content-Length: 0

-------------------------------------------------

Ed. 07

The 180 Ringing (or 183 Progress Session) is a

provisional message sent by the OXE, this

message is used to inform the caller, that the remote

party is ringing. This message can contain

SDP to provide the Ring back tone RBT), if no SDP, the

RBT must be played localy on the

system initator of the call.

43

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

 4) The SIP equipment receives a 200ok answer from the

OXE

Mon Jun 25 11:10:19 2012 SEND MESSAGE TO NETWORK

(172.27.141.210:63016 [UDP]) (BUFF LEN = 972)

----------------------utf8----------------------SIP/2.0
200 OK

Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER,

SUBSCRIBE, OPTIONS, UPDATE

Contact: sip:oxe-ov.alcatel.fr

Supported: replaces,timer,path,100rel

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

Session-Expires: 1800;refresher=uas

P-Asserted-Identity: "IPtouch 172.27.142.64"

<sip:[email protected];user=phone>

Content-Type: application/sdp

To: "31004" <sip:31004@oxe-

ov.alcatel.fr>;tag=bb28096d41c595340f577a538bf30d54

From: "31026" <sip:[email protected]>;tag=e9708b0f

Call-ID: MzI0MjQ4MmQ5NjMzZTVmZTlmYTE5NTVhMGNiZWI0ODQ.

CSeq: 1 INVITE

Via: SIP/2.0/UDP
172.27.141.210:63016;received=172.27.141.210;branch=z9hG4bK
d87543-88163a3aa534591a-1--d87543;rport=63016

Content-Length: 242

v=0

o=OXE 1340615417 1340615418 IN IP4 172.27.141.151

s=abs

c=IN IP4 172.27.142.64

t=0 0

m=audio 32514 RTP/AVP 18 101

a=sendrecv

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=ptime:20

a=maxptime:40

a=rtpmap:101 telephone-event/8000

-------------------------------------------------

The 200ok is used to open the SIP dialog (in that case),
when the called party hang up, the OXE

sends this 200ok with a SDP to provide the RTP

information for connection.

6) The SIP equipment sends a ACK to the OXE

Mon Jun 25 11:10:19 2012 RECEIVE MESSAGE FROM NETWORK

(172.27.141.210:63016 [UDP])

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----------------------utf8----------------------ACK
sip:oxe-ov.alcatel.fr SIP/2.0

Via: SIP/2.0/UDP 172.27.141.210:63016;branch=z9hG4bK-

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
d87543-342bae0b06436266-1--d87543-;rport

Max-Forwards: 70

Contact: <sip:[email protected]:63016>

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To: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=bb28096d41c595340f577a538bf30d54

From: "31026"<sip:[email protected]>;tag=e9708b0f

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Call-ID: MzI0MjQ4MmQ5NjMzZTVmZTlmYTE5NTVhMGNiZWI0ODQ.

CSeq: 1 ACK

User-Agent: SIP Phone

Content-Length: 0

-------------------------------------------------

Ed. 07

44

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The ACK is used to confirm the dialog. The ACK must

contain a SDP if there is no SDP on the

INVITE

7) The SIP equipment can send or receive a BYE, when the

call is stopped

1340615420 -> Mon Jun 25 11:10:20 2012 SEND MESSAGE TO

NETWORK (172.27.141.210:63016 [UDP]) (BUFF LEN = 454)

----------------------utf8----------------------BYE
sip:[email protected]:63016 SIP/2.0

Supported: replaces,timer,path,100rel

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

To: sip:[email protected];tag=e9708b0f

From: "31004" <sip:31004@oxe-

ov.alcatel.fr>;tag=bb28096d41c595340f577a538bf30d54

Call-ID: MzI0MjQ4MmQ5NjMzZTVmZTlmYTE5NTVhMGNiZWI0ODQ.

CSeq: 394681697 BYE

Via: SIP/2.0/UDP
172.27.141.151;branch=z9hG4bK07f995532987067a46049349aeaf05

Max-Forwards: 70

Content-Length: 0

-------------------------------------------------

The BYE is used to stop the dialog

8) The SIP equipment can send or receive a 200ok, to

confirm the BYE

Mon Jun 25 11:10:20 2012 RECEIVE MESSAGE FROM NETWORK

(172.27.141.210:63016 [UDP])

----------------------utf8----------------------SIP/2.0
200 OK

Via: SIP/2.0/UDP
172.27.141.151;branch=z9hG4bK07f995532987067a46049349aeaf05

Contact: <sip:[email protected]:63016>

To: <sip:[email protected]>;tag=e9708b0f

From: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=bb28096d41c595340f577a538bf30d54

Call-ID: MzI0MjQ4MmQ5NjMzZTVmZTlmYTE5NTVhMGNiZWI0ODQ.

 CSeq: 394681697 BYE

User-Agent: SIP Phone

Content-Length: 0

-------------------------------------------------

Ed. 07

45

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11. TROUBLESHOOTING

This section provides step-by-step instructions and

troubleshooting actions when you run into trouble.

When a SIP issue is present on the OXE, it is necessary

to find the cause of this trouble. To do this, it

necessary to make some investigations to find it.

Regarding the issue, different ways of investigations

are possible.

SIP call not possible

Voice problem

Fax transmission problem

DTMF issue

SIPMOTOR “crash”

...

Before to start, here some explainations about the

SIPMOTOR functionning and the traces in case of SIP

calls.

11.1

SIPMOTOR processes

The first step is to check if all the SIPMOTOR processes

are running well on the OXE.

For this, you can use the command “ps -edf | grep sip”,
can use the command “pidof sipmotor” to get all the

pid numbers.

In R9.1:

(1)OXE> ps -edf

root

2033

root

2139

root

2140

root

2141

root

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2142

| grep sip

822 0 Feb22

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2033 0 Feb22

2033 0 Feb22

2033 0 Feb22

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2033 0 Feb22

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?

00:00:00

00:00:07

00:00:00

00:00:00

00:00:00

/DHS3bin/servers/sipmotor

/DHS3bin/servers/sipmotor

/DHS3bin/servers/sipmotor

/DHS3bin/servers/sipmotor

/DHS3bin/servers/sipmotor

| grep sip

801 0 2011

2202 0 2011

2202 0 2011

2202 0 2011

2202 0 2011

00:00:00

00:00:00

00:00:00

00:00:00

00:00:00

[#sipmotor]

[sipmotor_tcl]

[sipmotor]

[sipmotor_dump]

[sipmotor_presen]

In R10.0

(1)OXE> ps -edf

root

2202

root

2203

root

2204

root

2205

root

2206

Ed. 07

46

TG0069

OmniPCX Enterprise

 Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

In normal functionning, the system displays the sipmotor

processes, we have 5 processes and the owner of

the processes is root (before the R9.1, the owner was

mtcl). According to the OXE release/version, the

number of processe can be different.

If the command gives you this result:

(1)OXE> ps -edf | grep sip

root

2033

822 0 Feb22 ?

root

2139 2033 0 Feb22 ?

mtcl

11942 10204 0 09:40 pts/0

00:00:00 /DHS3bin/servers/sipmotor

00:00:07 /DHS3bin/servers/sipmotor

00:00:00 grep sip

In that case, you don‟t have the good number of

processes, you can make a double bascul or a reboot the

CPU must be performed (shutdown -r 0).

If you run the command, and you have the next result:

(1)OXE> ps -edf | grep sip

root

2033

822 0 Feb22 ?

mtcl

12400 10204 0 09:53 pts/0

00:00:00 [#sipmotor <defunct>]

00:00:00 grep sip

In that case, the SIPMOTOR processes have been restarded

(automatically or manually), but the

configuration of the SIP is not well done, so the

configuration must be checked:

- The configuration of the SIP trunk group, used on the

local SIP gateway (node number,

etc…).

- The configuration of the local SIP gateway is well

done (good SIP trunk group used, etc…).

After modifications, the OXE must be rebooted (shutdown

-r 0).

11.2

SIPMOTOR memory used

When a problem is present on SIP, it is important to

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check the use of memory for the SIPMOTOR.

For this, run the command top -p “PID of the SIPMOTOR”.

10:35am up 3 days, 19:49, 1 user, load average: 9.00,

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9.00, 9.00

1 processes: 1 sleeping, 0 running, 0 zombie, 0 stopped

CPU states: 0.0% user, 0.0% system, 0.0% nice, 100.0%

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idle

Mem:

901304K av, 275124K used, 626180K free,

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0K shrd,

4752K buff

Swap: 1052216K av,

2180K used, 1050036K free

177596K cached

PID USER

27956 root

CLS PRI NI

FIFO 99 -12

SIZE RSS SHARE STAT %CPU %MEM

3996 3996 3616 S.<

0.0 0.4

TIME COMMAND

0:00 #sipmotor

The information to check are the “%CPU” and “%MEM”:

- If they are increasing when the traffic is more and

more higher and decreasing when the

traffic is going down, it seems that there is no issue

present about memory leak.

- If they are increasing continously, even if there is

no traffic, in that case a problem is

present, and a SR must be opened for analyse.

When memory leak is present, swap partition incidents

are also generated. If the next message is present,

check with the command top to see if the SIPMOTOR is

using to much memory.

20/03/12 15:15:24 000002M|---/--/-/---|=2:2071=Swap

partition 24 per cent full

Ed. 07

47

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.3

Check the SYSTEM and SIPMOTOR backtraces/alarms

11.3.1

Backtraces

“excvisu”

The excvisu can be used to see if system backtraces have

been generated by the OXE.

To know if the backtrace is about SIP, checks the next

information:

“SIPM”, it means that the backtrace is on the SIPMOTOR

itself.

==============================

There is a new exception. Its address is : 0XBFFFF118 in

SIPM. Monitel time : 024283. Date : Tue Apr

2010

 Application-exception no 11 in SIPM,
PC=0xbffff118:3221221656 --> _end

* SIPM Backtrace: 0x081631c8:135672264 -->

CResponse::create

* SIPM Backtrace: 0x08185ce0:135814368 -->

CTransProceedingState::createResponse

* SIPM Backtrace: 0x08152c09:135605257 -->

CTransaction::createResponse

* SIPM Backtrace: 0x0814d3bf:135582655 -->

CDialog::createResponse

* SIPM Backtrace: 0x0816ab94:135703444 -->

CCall::makeGenericResponse

* SIPM Backtrace: 0x080e8f8b:135171979 -->

CMotorCall::makeResponse

* SIPM Backtrace: 0x080e642e:135160878 -->

CMotorCall::emitServerFailureMessage

6 10:46:40

Backtrace for SIP Extension, the subtype information

contains “SIP_EXTENSION”

==============================

There is a new exception. Its address is : 0X092EEAA3.

Monitel time : 1961696. D

ate : Thu Feb 21 18:45:46 2008

Applicative-Error-Backtrace, thread 1371,

PC=0x092eeaa3:154069667, eqt=1380, ser

v=0 --> Kb_Interro

Eqt type=POS_NUM, cr=4, cpl=0, der_us=0, term=12,

subtype = SIP_EXTENSION

* Backtrace: 0x082f9c8e:137337998 EBP 0x01856e94 -->

egzis_li

* Backtrace: 0x08ae2b6b:145632107 EBP 0x01856ea8 -->

testprio

* Backtrace: 0x09263051:153497681 EBP 0x01856ec0 -->

real_main

GS=0000055b

FS=00003039

ES=00000000

DS=0000002b

EDI=09823194 ESI=00000000 EBP=01856e94 ESP0=00000000

EBX=538c4e97 EDX=538ca920 ECX=210b3100 EAX=00000019

EIP=092eeaa3

CS=00000023 ESP3=01856e78 EFLAGS=00000246

Continuing at previous PC=0x092eeaa3:154069667

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Ed. 07

If the address start by “cr=19” the backtrace can be

linked to the SIP Trunk Group, the cr=19

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corresponds to the virtual shelf for the IP-Link, so the
Backtrace could be for another feature

using the IP-Link, use the command “trkvisu” to see if

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the position (“cr” + “cpl” + “term”)

corresponds to the SIP Trunk Group.

48

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TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

==============================

There is a new exception. Its address is : 0X093C1E26.

Monitel time : 250434. Date : Tue Mar 20 09:18:48 2012

Application-exception no 5, thd 1176,

PC=0x093c1e26:154934822, eqt=13517, serv=0 --> __CHECK__

Eqt type=JONCT, cr=19, cpl=0, der_us=0, term=2

* Backtrace: 0x08333369:137573225 EBP 0x01826db8 -->

nuphmult

* Backtrace: 0x08990328:144245544 EBP 0x01826ddc -->

process_ccbs_exec_poss

* Backtrace: 0x08999135:144281909 EBP 0x01826e30 -->

analyse_facilite_abc

* Backtrace: 0x08999a05:144284165 EBP 0x01826e3c -->

analyse_facilite

* Backtrace: 0x087fc81d:142592029 EBP 0x01826e4c -->

arr_ipns

* Backtrace: 0x08836851:142829649 EBP 0x01826e7c -->

sui_arr_q931

* Backtrace: 0x08836b09:142830345 EBP 0x01826eac -->

arr_q931

For all backtraces about SIP, a SR must be opened for

R&D analyses.

“sipmotor.crash”

Under /tmpd, there is a file called “sipmotor.crash”

containing the SIPMOTOR “crash” information (file

includes on the Infocollect).

(1)OXE> more /tmpd/sipmotor.crash

sipmotor.crash generated at Tue Oct 19 09:15:42 2010

1287472542 -> [CMotorCallManager::insertCallwithEqt]

CMotorCall 1911 inserted.4NzQyZjY2NTI2ZT

1287472542 -> [quoteString] => "31017"onse]Trying to

find the right dialogte = Terminated, cu

1287472542 ->
1186[CMotorCall::inviteBuildFromAssertedId] no
P_Asserted_Identity c33435cb1ed7

1287472542 -> 1186[CMotorCall::setFilterUsedMode] To be

traced = 0undterminated reason : None

If the sipmotor.crash file increase after SIP calls, to

see which calls are causing this, make SIPMOTOR

traces, all the information present in this file, are

taken from the SIPMOTOR, and seen on the traces.

11.3.2

Alarms

On the OXE, some SIP incidents can be generated, next,

the explanation of each one.

5800: “X” SIP trunk group put into service.

 This incident is used to informed you that the SIP trunk

group “X” is put in service.

5801: “X” SIP trunk group put out of service.

This incident is used to informed you that the SIP trunk

group “X” is put out of service, if the trunk group is

put out of service automaticaly by the OXE (without

human action) open a SR for analyses.

5812: SIP external gateway “Y” is in service.

This incident is used to informed you that the SIP

gateway “Y” is in service.

5813: SIP external gateway “Y” is out of service.

This incident is used to informed you that the SIP

external gateway “Y” is put out of service, if the
external

SIP gateway is put out of service automaticaly by the

OXE (without human action) open a SR for analyses.

The state of the SIP Trunk Group and the external SIP
gateway are linked:

Ed. 07

49

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

If the SIP Trunk Group associated to the SIP external

gateway is out of service, the SIP

external gateway is out of service too.

If the external SIP gateway is out of service, the SIP

trunk group associated is out of service

also, except if this SIP Trunk Group is associated to

another external SIP gateway, and this

one is in service.

If all the external SIP gateway associated with one SIP

Trunk Group are out of service, the

SIP Trunk group will be out of service.

5814: Critical failure in SIP component.

5815: Major failure in SIP component.

5816: Minor failure in SIP component.

These 3 incidents give an information about a problem

during SIP exchanges (Registrations, Calls, etc...).

To get more information about thes incidents, go under

/tmpd/ and open the sipalarm files.

(1)cpua_ov> ll sipal*

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-rw-rw-rw1 root

-rw-rw-rw1 root

-rw-rw-rw1 root

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-rw-rw-rw1 root

-rw-rw-rw1 root

-rw-rw-rw1 root

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-rw-rw-rw1 root

-rw-rw-rw1 root

-rw-rw-rw1 root

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-rw-rw-rw1 root

tel

root

tel

tel

root

root

root

root

root

root

15658

20456

20529

20529

20553

20553

20553

20553

20553

20553

Feb

Nov

Nov

Nov

Nov

Oct

Oct

Oct

Oct

Oct

23

10

10

10

30

30

31

31

31

09:54

11:48

12:30

13:28

09:17

15:29

23:47

07:16

15:38

 23:59

sipalarm.log

sipalarm1.log

sipalarm2.log

sipalarm3.log

sipalarm4.log

sipalarm5.log

sipalarm6.log

sipalarm7.log

sipalarm8.log

sipalarm9.log

The file sipalarm.log corresponds to the current one.

To make the link between the incident and an entrie in

the sipalarm file, check the date and hour of the

incident generation with incvisu:

01/14/11 15:46:02 000001M|---/--/-/---|=2:5816=Minor

failure in SIP component

After check on the sipalam file the entry in this time:

> 01/14/11 - 15:46:02

Minor alarm

[receiveInviteEvent] Call: eqt: 1674 INITIAL_STATE

failed to emit an Invite message.

In that case, the SIPMOTOR was not able to send an

INVITE (lake of licenses for instance).

When the incidents 5814, 5815 and 5816 are generated,

and if you constat some problems on the OXE, a

SR can be open with the information from the command

incvisu and the sipalarm files (or send the

Infocollect).

Ed. 07

50

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.4

SIP traces

The OXE has different levels of traces to get

information from the different elements (SIPMOTOR, Call

handling, IP).

The traces can be run on the Main CPU and on the Stand-
By CPU.

11.4.1

SIPMOTOR traces

The SIPMOTOR traces are used to make traces at the

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sipmotor level, for this, the command “motortrace”

can be used to set the level of trace you need.

motortrace (v5.2.0) verbosity = 00000000

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Correct usage is:

motortrace trace-level To set the current trace level.

motortrace +T_TRACE

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To add a single level to the current trace.

motortrace -T_TRACE

To remove a single level to the current trace.

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T_MOTOR, T_SIP, T_PKT_IN, T_PKT_OUT, T_IPC_IN,
T_IPC_OUT, T_INTERNAL_DESTR,

T_LOG, T_DEBUG, T_FW, T_DB, T_US, T_MOTOR_TEST,

T_TRANSPORT, T_ADNS

trace-level :

0 : No trace (only Alarm)

1 : Basic trace (T_PKT_IN|T_PKT_OUT|T_IPC_IN|T_IPC_OUT)

2 : Medium trace
(T_MOTOR|T_SIP|T_PKT_IN|T_PKT_OUT|T_IPC_IN|T_IPC_OUT)

3 : All traces

4 : Medium trace dupli

(T_MOTOR_TEST|T_SIP|T_PKT_IN|T_PKT_OUT|T_IPC_IN|T_IPC_OUT)

5 : All traces + dupli

6 : All traces + T_TRANSPORT + T_ADNS

7 : All traces + T_INTERNAL_STRUCT

8 : Medium trace options

(T_MOTOR|T_SIP|T_PKT_IN|T_PKT_OUT|T_IPC_IN|T_IPC_OUT|T_OPTI

9 : All traces + options

c : Configuration

Traces will be directed to the window, where traced is

executed (TL).

Current level of trace is:

sipmotor

trace-level

The

“trace-level”0 is(No

thetrace).

most used options for motortrace traces, the other are

mostly used by the R&D

(if

needed).

According to the level of traces, the information given

are different.

If you select 0, that means you have no SIP traces, only

the alarms are displayed

If you select 1, we have only the SIP messages and the

information given by the

Call Handling

If you select 2, we have more information given,

compares to the level 1, we can

 see the SIPMOTOR checking the external SIP gateway

associated to the INVITE

received (for instance)

If you select 3, we have all the SIP traces, this level

is the most used.

If you select 4, we have the level 2 traces + the

duplication information (SIP

exchanges between the Main and the Stand-By CPUs)

If you select 5, we have the level 3 traces + the

duplication information (SIP

exchanges between the Main and the Stand-By CPUs)

If you select 6, we have all the traces + the transport

trace (network) + the DNS

information

If you select 7, we have all the traces + the internal

structure of SIP in SIPMOTOR

(From R10.0)

If you select 8, we have the level 2 traces + options

(From R10.0)

If you select 9, we have all the traces + all options

(From R10.0)

When you increase the level for the traces, you increase
also the size of the traces.

The command “traced” is used to output the traces, some

options are possibles:

Ed. 07

51

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

If you use the “traced &”, the trace is running in

background.

If you use the “traced >/tmpd/name_of_the_file.log” to

put the trace in a file.

If you use the “traced -1 /tmpd/traced -s 10000000 -f 50

-d”, to make rotating trace.

(1)OXE> motortrace 3

motortrace (v5.2.0) verbosity = 0037b524

sipmotor trace-level set 3 (All traces).

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(1)OXE> traced

** UNIX-trace-daemon started ... (static user group No

1) **

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traced started ...

Make a “CTRL + C” to stop the trace or “killall traced”

when the trace is running in background.

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The level of traces must be put back motortrace 0 after
traces taken to avoid memory leak.

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The option “c” can be used to display all the SIP
configuration (local)

(1)OXE> motortrace c

motortrace (v5.2.0) verbosity = 0037b524

sipmotor trace-level set c (data dump).

Proxy parameters.

=================

sip stack version

4.0.006.022

initial_timeout

500

timer_t2

4000

recursion

min_auth_method

0 NONE=0 DIGEST=2

auth_realm

cpua

sipDnsTimerPrimSecond 5000

onlyAuthIncomingCalls 1

quarantine and trusted addresses:

nb_msg_by_period

25

period

framework_quarantine_period 1800

Gateway parameters.

===================

url_install

url_gw

url_hostname

num_ss_reseau

num_faisc

proxy_address

DNS_localDomName

DNS_type

DNS_primaire

DNS_secondaire

prack_required

out_proxy

proxy_port

proxy_transport

sipSubsMinDuration

sipSubsMaxDuration

sipSessionTimer

sipMinSessionTimer

SessionTimerMethod

sipCac

SDP_in_180

 sip_info_enable

payload

seplos

172.27.141.151

oxe-ov

10

not used

alcatel.fr

0 dnsa=0, dnssrv=1

Unused

Unused

0 AUCUN=0 INTEGRE=1 EXTERNE=2

5060

1 TCP=0 UDP=1

1800

86400

1800

900

1 re-invite=0, update=1

97

...

Ed. 07

52

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.4.2

Call Handling traces

Call Handling traces can be provided in case of issue,

because there is a permanent link between the Call

Handling and the SIPMOTOR, so the call can be rejected

by the Call Handling and not from the SIPMOTOR.

The SIPMOTOR traces and the Call Handling traces must be

done simultaneously.

Here, the basic Call Handling trace done on the OXE.

(1)OXE> tuner km

(1)OXE> tuner all=off

(1)OXE> tuner clear-traces

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(1)OXE> trc i

+--------+-------+--------+--------+---------+---------
+----------+------+

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| filter | desti | src_id | cr_nbr | cpl_nbr | us_term |
term_nbr | type |

+--------+-------+--------+--------+---------+---------
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+----------+------+

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|

+--------+-------+--------+--------+---------+---------
+----------+------+

(1)OXE> tuner +cpu +cpl +at +time hybrid=on

(1)OXE> actdbg all=off

Thu Feb 24 10:41:42 CET 2011

(1)OXE> actdbg sip=on

Thu Feb 24 10:41:52 CET 2011

(1)OXE> mtracer -a

Traces Analyser activated

mtracer started ...

(858432:000001) MTRACER host (172.27.141.149, OXE),

version: R9.1-i1.605-23-fr-c0

(858432:000001) MTRACER num: 002, time: 2011/02/24

10:42:16, loss: 0%

After, according to the issue, it is possible to add

options for traces. For instance, if from a SIP device
you

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are not able to dial an ARS prefix, you can add “ars=on”
on the actdbg. The Call Handling traces must be

adapted to the issue.

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Here an example of trace asked by R&D:

(1)OXE>

(1)OXE>

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(1)OXE>

(1)OXE>

(1)OXE>

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(1)OXE>

(1)OXE>

(1)OXE>

(1)OXE>

(1)OXE>

tuner km

tuner clear-traces

tuner all=off

trc init

actdbg all=off

tuner +at +tr +xtr +s

tuner +cpu +cpl

tuner hybrid=on

actdbg sip=on csip=on fct=on isdn=on abcf=on ext=on

rtp=on cnx=on comp=on voip=on ccdn=on cstarout=on

mtracer -a -u -g

Ed. 07

53

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Three actdbg options are linked to SIP:

 sip, corresponding to sip (globals SIP traces).

 csip, corresponding to the SEPLOS terminals.

 nsip, corresponding to NOE-SIP terminals (From R10.0).

11.4.3

Tcpdump / Network traces

The tcpdump or network traces can be done as well, to

check if the problem is from the network or the

network layer of the CPU. Tcpdump must be run under root

account.

The network traces are very usefull when you have issue
about one way call, DTMF, FAX, etc…

The tcpdump or network traces must be done

simultaneously with the SIPMOTOR and the Call

Handling traces.

(1)OXE> su root

Password:

[root@OXE tmpd]# tcpdump -s 2000 -w trace.cap

tcpdump: listening on eth0, link-type EN10MB (Ethernet),

capture size 2000 bytes

Runing the tcpdump with the option “-s 2000” and the
option “-w trace.cap” is used to be able to open this

trace with wireshark (http://www.wireshark.org/).

Rotating trace can be used with the next syntax:

[root@OXE tmpd]# tcpdump -C 10 -w /tmpd/mytcpdump.cap -W

10 -s 2000 &

-C corresponds to the size of the file (10 corresponds

to 10 Megabytes)

 -W corrsponds to the number of files

More options are available.

Ed. 07

54

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.5

Mantenance commands

This chapter explains all the SIP maintenance commands

available on OXE.

11.5.1

sip

===========================================================

T O O L S

A V A I L A B L E

F O R

S I P

P U R P O S E

===========================================================

trkstat

trkvisu

sipacces

: Shows the trunks states in a trunk group

: Shows the trunks parameters in a trunk group

: Shows the SIP trunk group numbers and the related

accesses

sipgateway

sipdump

: Shows the main SIP gateway parameters

: Shows the main SIP gateway internal resources

sipextgw

sippool

: Shows the external SIP gateways parameters

: Shows the external SIP gateways membership of pools

sipdict

: Shows the SIP dictionnary records

sipauth

: Shows the SIP authentification records

sipregister : Shows the SIP end points IP address

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registered

csipsets

: Shows the list of configured SIP extension

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csipview com : Shows the list of SIP extension in
communication

csiprestart : Reset the dynamic datas (CH + CC) of

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blocked SIP extension

sipextusers

: Shows the SIP devices with gateway

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The command “sip” gives all the commands related to SIP.

11.5.2

trkstat

+==========================================================

S I P

T R U N K

S T A T E

Trunk group number : 10

Trunk group name

: SIP_local

Number of Trunks

: 62

+-------------------------------------------------------
-----------------------+

Index :

10

11

12

13 |

State :

 F |

+-------------------------------------------------------
-----------------------+

Index :

14

15

16

17

18

19

20

21

22

23

24

25

26 |

State :

F |

+-------------------------------------------------------
-----------------------+

Index :

27

28

29

30

31

32

33

34

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35

36

37

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38

39 |

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State :

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F

F |

+-------------------------------------------------------
-----------------------+

Index :

40

41

42

43

44

45

46

47

48

49

50

51

52 |

State :

F |

+-------------------------------------------------------
-----------------------+

Index :

53

54

55

 56

57

58

59

60

61

62

State :

+-------------------------------------------------------
-----------------------+

| F: Free

B: Busy

Ct: busy Comp trunk

| Cl: busy Comp link

| WB: Busy Without B Channel|

Cr: busy Comp trunk for RLIO inter-ACT link

| WBD: Data Transparency without chan.| WBM: Modem

transparency without chan. |

| D: Data Transparency

M: Modem transparency

+-------------------------------------------------------
-----------------------+

Ed. 07

55

TG0069

OmniPCX Enterprise

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TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

The command “trkstat” + SIP Trunk Group number gives the

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B channels used on the SIP Trunk group

assocaited to a gateway.

11.5.3

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trkvisu

****************** data in Trunk_Group structure

****************

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********

data REMOTE TRUNK

TrunkName

= SIP_local

discrLogId

= -1

ton_a_used = 0

em_repfix = 0

privLine

= 0

reservop

= 0

reserauto = 0

Trunksearchs = 0

ftranscom

= 0

frondier = 0

typTrunk :

(6) => T2-SIP

Next_Trunk

= -1

nbdigitsem

= 0

tab_proto = -1

var IPNS = 1

node_number

= 1

network_number = 10

trunk_reg_sig = 0

special_it_par_quantum = 1

cat_restrictionService_in = 10,
cat_restrictionService_out = 10

Priority ===> Level= 0, Mode= 0, Preemption= 0

mpt1343 = 0, callbackTrunkbusy = 1 rerouting = 0

********

data Link

cat_signa = 31 ch_channelb = 1 overflow_it = 1

access_turn = 1 network_mode = 0

+-------------------------------------------------------
------------------------------------|

ocupjonc for SIP TG

+-------------------------------------------------------
------------------------------------| SIP Trunk group
information on TX side

| i = 0, min = 0, max = 62

| (num_crist - num_cpl - num_term) = (19-0-1)

| last it used = 0, monlap = 30, network_mode = 0

nbr_trunk_created = 62

| nbr_trunk_busy : start = 0 arrived = 0 mixed = 0

| it_reserved

 : start = 0 arrived = 0 mixed = 62

| it_max_Q0

: Start = 0 arrived = 0 mixed = 62

| it_max_Q1

: Start = 0 arrived = 0 mixed = 0

| access_level2 = CONNECT2

+-------------------------------------------------------
------------------------------------| outservice | res |
Busy | nulog |trans|neqtdyn|E64 RN64 EN64| OVPN | neqph
| adr

+-------------------------------------------------------
------------------------------------| FREE

| no | free | 5001 |

1 |

-1 | 0

0 | 0

2314 |SIP Trunk 1

...

| FREE

| no | free | 5062 |

1 |

-1 | 0

0 | 0

2376 |SIP Trunk 62

+-------------------------------------------------------
------------------------------------+-------------------
--------------------------------------------------------
----------------| SIP Trunk group information on GX side

| (num_crist - num_cpl - num_term) = (19-0-0)

| monlap = 29, mode_reseau = 1 nbr_jonc_cree = 62

| Trunks from nulog 5063 (neqph 2250) to nulog 5124

(neqph 2312)

+-------------------------------------------------------
------------------------------------index_max = 125 ;
nbjonc = 62

cristal

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4

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7

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10

11

LastTrunkused =

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0

cristal

12

13

14

15

16

17

18

19

LastTrunkused =

62

LastTrunkUsed common = 0

idx_rfo = 0

channel_reserv = 0

dert0_used = 0 dert0mixt_used = 0

dert0wo_used = 0

********

data TRUNK_LOCAL

a_paying

= 0

Trunkdisa

= 0

itpermnt

= 1 trans_num = 0

tr_q23

= 0

reach_boss = 0

secretcode = 0 ach_film = 1

accesscode

= 0

 gp_d_Hold

= 0

categ_ptt = 31 blf etat = 1

entity_nr

= 0

nb_digit_used = 0

Trunkdissu

= 0

dto_about = 0

reused_channelb = 0

number_to_be_added :

mode_ddi

= 0

refptt = /

nbchminp = 0

x25used

= 0

vpnRate

= 50

vpnCostLimit = 0

immTrkForVpn = 1

businessPercent = 0

nbACDCall

= 0

tax_nds

= 1

send_prog = 1

ip_qual_prof = Profile #1

t2spec

= S_SIP

compression_type = 0 (0: Default [ie : G729], 1 : G711)

d_channel_hyb = 0

Ed. 07

56

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

The information given are the same compares to a

“normal” T2 access, this command can be usable to find

the equipment of a SIP Trunk Group, or the “neqt”.

A SIP Trunk group has two “sides”, the TX (USER) and GX

(NETWORK). When a call is done on a SIP

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Trunk Group, the call is leaving on the SIP TG and comes
back on the same SIP TG, it is like an internal SIP

loop.

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
11.5.4

sipaccess

+-------------------------------------------------------
(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
-----------------------+

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
|

SIP Trunk Group Access

+-------------------------------------------------------
-----------------------+

| TG Nb |

10

12

11

186

187

| Access | User - Net | User - Net | User - Net | User -

Net | User - Net |

+-------------------------------------------------------
-----------------------+

30 - 29

33 - 32

35 - 34

37 - 36

39 - 38

. . .

41 - 40

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
. . .

. . .

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
|

. . .

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
. . .

. . .

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
|

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

. . .

10

. . .

. . .

. . .

. . .

. . .

11

. . .

. . .

. . .

. . .

. . .

12

. . .

. . .

. . .

. . .

. . .

13

. . .

. . .

. . .

. . .

. . .

14

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
. . .

. . .

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
|

. . .

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
. . .

. . .

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|

15

. . .

. . .

. . .

. . .

. . .

16

. . .

. . .

. . .

. . .

. . .

+-------------------------------------------------------
-----------------------+

The command “sipacces” gives the access numbers used for

each SIP TG.

In that case, for the TG number 10 with 2 accesses

managed, the OXE uses the accesses 30 for TX and 29

for GX, these accesses numbers can be found with the

command “trkvisu” (search for “monlap”).

In that case, for the TG number 12 with 4 accesses

managed, the OXE uses the accesses 33 and 41 for TX

then 32 and 40 for GX.

11.5.5

sipgateway

+-------------------------------------------------------
----------------+

SIP Gateway

+-------------------------------------------------------
----------------+

Machin name

: oxe-ov

IP Address

 : 172.27.142.53

Subnetwork number

SIP Trunk Group

: 10

: 10

DNS Informations :

DNS local domain name

: alcatel.fr

+-------------------------------------------------------
----------------+

Trusted IP Address List

+-------------------------------------------------------
----------------+

+-------------------------------------------------------
----------------+

Quaranted IP Address List

+-------------------------------------------------------
----------------+

Ed. 07

57

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The command “sipgateway” gives the information about the

local SIP configuration.

The next information are diplayed:

11.5.6

Machine name corresponds to the “nodename” managed under

netadmin.

IP address corresponds to the main IP address of the

main CPU.

Subnetwork number correponds to the network associated

to the local SIP gateway.

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SIP Trunk Group correponds to the SIP TG associated to
the local SIP gateway.

DNS local domain name correponds to the DNS suffix

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
managed on the local SIP gateway.

Trusted IP Addresses List corresponds to the IP

addresses managed on the “Trusted IP

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Addesses”

Quaranted IP Addresses List corresponds to the IP

addresses managed on the

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“Quarantined IP addresses”.

sipdump

The sipdump tool gives some information about SIP calls

and the SIP gateway. It‟s useful in order to know

which states have the SIP calls, which are handled by

the SIP gateway, to release a call, which is inactive,

etc…

It allows defining some filters in order to display the

traces of SIP calls, according to SIP calls
characteristics

(“From”, “To”, “P_Asserted”, “Request URI” headers).

Activation:

Set a trace level very low (set by motortrace – lowest

trace level by motortrace 0), and disable filters.

Run the “traced &” command.

Run the command “sipdump”.

For better view, run “sipdump” and “traced” in different

telnet sessions.

A “Call” corresponds to a SIP voice call, but also for a

subscription, notify, etc…

Sometimes, choices must be done twice to get the

outputs.

R10.x

R9.1

SIP Gateway resources menu

SIP Gateway resources menu

1 - Dump the gateway management datas

2 - Dump a call

3 - Display the number of calls

4 - Display the calls-neqt mapping

5 - Display the calls list

6 - Display the detailed calls list

7 - Release a call

8 - Display subscription list

9 - Display calls through a gateway

10 - Display calls in a trunk group

11 - SIP traces filters

12 - Display registred users

13 - Display CPU-SSM connections

14 - Display memory allocation

15 - Display IP cache from ext gw

0 - Exit

Ed. 07

10

58

Dump the gateway management datas

Dump a call

Display the number of calls

Display the calls-neqt mapping

Display the calls list

Release a call

Display subscription list

Display calls through a gateway

Display calls in a trunk group

SIP traces filters

Exit

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

1 – Dump the gateway management datas :

Wed

Wed

Wed

Wed

Wed

Wed

Mon

Mon

Wed

Wed

Wed

Jan

Jan

Jan

Jan

Jan

Jan

Jun

Jun

Jan

Jan

Jan

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4

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4

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4

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4

14:48:42 2012

Gateway Management Datas

14:48:42 2012 -----------------------------------------

-14:48:42 2012

14:48:42 2012 Use of licences

: Yes

14:48:42 2012 Number of initial licenses

: 20

14:48:42 2012 Number of available licences

: 15

12:48:42 2012 Number of initial Tls licenses

: 5

12:48:42 2012 Number of available Tls licences : 5

14:48:42 2012

14:48:42 2012 Main server

: Yes

14:48:42

2012 server

Degraded

mode

: Yes

Main

corresponds

to the role of the

CPU (Main

or Stand-By).

Use of licenses means that the OXE is using SIP, license

point of view.

Number of initial licenses corresponds to the number of

licenses bought.

Number of available licenses corresponds to the number

of licenses remaining. The difference

with the Number of initial licenses give the number of

licenses used when this choice is done.

Number of initial Tls licenses corresponds to the number

of licenses bought for TLS.

Number of available Tls licenses corresponds to the

number of licenses remaining for TLS. The

difference with the Number of initial Tls licenses give

the number of licenses for TLS used when

this choice is done.

Main server gives the role of the CPU where you run the
“sipdump” command.

Degraded mode is used when the SIPMOTOR reaches a

 threshold of SIP contexts treatment, in

that case, the SIPMOTOR switches in degraded mode to

reject all the incoming SIP messages by

a 503 response, with a "Retry-After" header, is sent to

the UAC. This is used to avoid SIPMOTOR

crash.

2 – Dump a call

Enter the “Neqt” of the SIP equipment + “Dialogid”, to

know them, use the choice 4 before.

1325686751 -> Wed Jan 4

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:18:56 2012

Wed Jan 4 15:19:07 2012

Wed Jan 4 15:19:07 2012

Wed Jan 4 15:19:07 2012

Wed Jan 4 15:19:07 2012

Wed Jan 4 15:19:07 2012

Wed Jan 4 15:19:07 2012

Wed Jan 4 15:19:07 2012

Wed Jan 4 15:19:07 2012

Wed JanEd.40715:19:07 2012

Wed Jan 4 15:19:07 2012

Wed Jan 4 15:19:07 2012

15:18:56 2012 ------------------------------------------

Call Dump

------------------------------------------Neqt

: 968-1

Call ID

: [email protected]

Current state

: COMPLETED_STATE

From

: sip:[email protected];user=phone

To

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
: sip:[email protected];user=phone

External VM:

: FALSE

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Sip Device:

: FALSE

Ext. Gateway

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
: Not used

Session Timer

: INVITE method

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--------------------------------------------------------
----------------------------Neqt - Call mapping

------------------------------------------Active Calls
(1 / 1)

Eqt =

968 dialogId = 1 <-> Call ID =

[email protected]

State = COMPLETED_STATE

59

Unactive Calls (0 / 1)

-------------------------------------------

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The “Current state”corresponds to the status of the

call:

 PROCEEDING_STATE : the call is in progress (ringing

for instance).

 COMPLETED_STATE : the call is established.

 TERMINATED_STATE : the call is finished.

From and To correspond to the caller and the callee.

External VM : False means that it is not an external SIP

Voice mail.

Sip Device: False means a SIP extension user (SEPLOS).

Ext. Gateway correponds to the external SIP gateway used

for the call.

Session Timer corresponds to the method used for it

according to the local SIP gateway

management:

 UPDATE Method: use UPDATE message to refresh the

session.

 INVITE method: use RE_INVITE message to refresh the

session.

Active Calls correponds to the SIP calls established

 Only COMPLETED_STATE is visible.

Unactive Calls corresponds to the SIP calls over or in

progress:

 Unactive + PROCEEDING_STATE, corresponds to a SIP call

in progress.

 Unactive + TERMINATED_STATE, corresponds to a SIP call

over, but its SIP

 context is still present on the SIPMOTOR. The maximum

duration of the context in

the SIPMOTOR is 32 seconds, during this period, the

SIPMOTOR will delete it. If

the SIP call context is still present after this delay,

the SIPMOTOR will not be able to

remove it by itself, a restart of the SIPMOTOR must be

done.

When a restart of the SIPMOTOR is done, all the SIP call

context will be lost, that means that

the calls are not anymore known by the SIPMOTOR.

3 – Display the number of calls

1325752599 -> Thu Jan

5 09:36:39 2012 stack data.

Thu Jan 5 09:36:39 2012

Thu Jan 5 09:36:39 2012

Thu Jan 5 09:36:39 2012

Thu Jan 5 09:36:39 2012

Thu Jan 5 09:36:39 2012

Thu Jan 5 09:36:39 2012

Thu Jan 5 09:36:39 2012

totalPutinBlackList…

==========

Calls :

Dialogs :

Transactions :

Requests :

Response :

DNS requests :

current

current

current

current

current

current

(4 max) / 59052 total

(6 max) / 59083 total

(6 max) / 59240 total

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
/ 59301 total

/ 309 total

(0 max) / 0 total / 0 foundInCache / 0 failed / 0

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Corresponds to the number of SIP calls, but also SIP
dialogs, SIP transactions, etc…

4 - Display the calls-neqt mapping.

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Thu

Thu

Thu

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Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Jan 5 10:19:59

Jan 5 10:19:59

Jan 5 10:19:59

Jan 5 10:19:59

Jan 5 10:19:59

Jan 5 10:19:59

Jan 5 10:19:59

Jan 5 10:19:59

Jan 5 10:19:59

JanEd.50710:19:59

Jan 5 10:19:59

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

------------------------------------------Neqt - Call
mapping

------------------------------------------Active Calls
(1 / 1)

Eqt =

968 dialogId = 2 <-> Call ID =

[email protected]

State = COMPLETED_STATE

Unactive Calls (0 / 1)

60

-------------------------------------------

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

Corresponds to the Active and Unactive calls present on

SIPMOTOR, for the sipdump choice 2,

it is necessary to have the “Neqt” and the “dialogid”,

here we have them for each call.

 5 - Display the calls list.

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Jan

Jan

Jan

Jan

Jan

Jan

Jan

Jan

Jan

Jan

Jan

10:25:54

10:25:54

10:25:54

10:25:54

10:25:54

10:25:54

10:25:54

10:25:54

10:25:54

10:25:54

10:25:54

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
-

2012

2012

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
2012

2012

2012

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
2012

2012

2012

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2012

2012

2012

------------------------------------------List of Calls

------------------------------------------Active Calls
(1 / 1)

Call ID =
[email protected]

State = COMPLETED_STATE

Unactive Calls (0 / 1)

-------------------------------------------

List the Active and Unactive SIP calls on the SIPMOTOR.

Recommended in case of licence

consumming issue.

6 - Display the detailed calls list.

Thu Jan 5 10:29:41 2012

Detailed list of Calls from Stack

Thu Jan 5 10:29:41 2012 --------------------------------

----------Thu Jan 5 10:29:41 2012 102 [CCallManager]
Dump - 1 CCall instance(s)

[1137] Call ID :
[email protected]

CCall 1137

Call-ID

: [email protected]

isClosed

: no

onlyInitialDialog

: no

==========================================================

InitialDialog client :

-------------------CDialog 1537

isClosed

: no

isProxy

: no

isRouted

: no

State

: Initial

Initial method : INVITE

Session-Timer :

isProxy

: no

supported

: I support

Min-SE

: 900

Session-Expires : 1800

 Refresher

: I refresh

Warning timer

: stopped

Session timer

: stopped

Refresh method :

Route set

: Contact :
sip:[email protected]:36128;rinstance=98cedca3f085d785

Messages :

---------------------------------------out:INVITE
[2012/01/05 10:19:54 CET]

in:180 (INVITE) [2012/01/05 10:19:54 CET]

in:200 (INVITE) [2012/01/05 10:19:55 CET]

--------------------------------------------------------
--------------------------------------Transactions :

-----------CTransaction 2138

State

: Proceeding

isClient

: yes

isCancelable

: no

isRouted

: no

isProxy

: no

Initial request

: INVITE (38)

Last response

: 180 (6)

Final response

: None

Ed. 07

61

Ack request

: None

Timers in progress : None

--------------------------------------------------------

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

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Session Iniation Protcol (SIP)

This choice is used to view the different exchanges

details for the SIP transactions.

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For each transaction, we have 3/4 groups of information
(3for call in progress, 4 for

established/closed):

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

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

SIP call information with the Call ID and the state of

the call:

- Closed

(isClosed= yes)

- In progress (onlyInitialDialog=yes)

- Established (isClosed= no and onlyInitialDialog=no)

InitialDialog client:

- This part corresponds to the information on the SIP

message received or

sent to establish a SIP transaction (INVITE, SUBSCRIBES,

etc…).

Transaction:

- This part corresponds to the status of the transaction

itself (type of

transaction, last message, etc…).

Dialogs:

- This part corresponds to the dialog information.

7 - Release a call.

- Enter the “Neqt” number and the “DialogId”, use the

choice 4 to find them.

Thu Jan

Thu Jan

Thu Jan

Thu Jan

Thu Jan

Thu Jan

Thu Jan

Thu Jan

Thu Jan

Thu Jan

Thu Jan

Thu Jan

eqt: 968

Thu Jan

5 12:05:45 2012 ----------------------------------------

--5 12:05:45 2012

Neqt - Call mapping

5 12:05:45 2012 ----------------------------------------

--5 12:05:45 2012

5 12:05:45 2012 Active Calls (1 / 1)

5 12:05:45 2012

Eqt =

968 dialogId = 1 <-> Call ID =

[email protected]

5 12:05:45 2012

State = COMPLETED_STATE

5 12:05:45 2012

5 12:05:45 2012

5 12:05:45 2012 Unactive Calls (0 / 1)

 5 12:05:45 2012 ----------------------------------------

--5 12:05:51 2012 ALARM: [receiveSuccessfulEvent] Call:
[email protected]

TERMINATED_STATE failed to emit a Successful message.

5 12:05:51 2012 ALARM: CPU main

An incident 5816 is seen on the OXE and the alarm is

visible on the sipalarm files.

8 - Display subscription list.

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Jan

Jan

Jan

Jan

Jan

Jan

Jan

Jan

Jan

Jan

12:11:33

12:11:33

12:11:33

12:11:33

12:11:33

12:11:33

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
12:11:33

12:11:33

12:11:33

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
12:11:33

2012

2012

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
2012

2012

2012

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
2012

2012

2012

2012

2012

------------------------------------------sipmotor
Subscription Map

key

[email protected]@message-summary

call no 1153

call Id NTUyZjA1ZmFiYTQ1MDI3N2U2ZTE1NzFkY2ZjZmM2MmQ.

delay

3600

------------------------------------------Number of
Subscription (s) : 1

end

Subscription

Mapin case of voice mail, for instance

- of

Thesipmotor

subscription

can be used

-------------------------------------------

to be able to be

notified if a message has been deposited on the voice

mailbox.

9 - Display calls through a gateway.

- Enter the External Gateway number.

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Jan 5 13:41:14

Jan 5 13:41:14

Jan 5 13:41:14

Jan 5 13:41:14

Jan 5 13:41:14

Jan Ed.507

13:41:14

Jan 5 13:41:14

Jan 5 13:41:14

2012

2012

2012

2012

2012

 2012

2012

2012

------------------------------------------Call ID

: [email protected]

Current state

: COMPLETED_STATE

From

: sip:32000@toto;user=phone

To

: sip:[email protected];user=phone

Session Timer

: UPDATE method 62

------------------------------------------Number of
Calls through this Gateway (151) : 1 (Active calls: 1)

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

10 - Display calls in a trunk group.

- Enter the SIP trunk group number (ISDN or ABCF).

Trunk Group Number : 10

Display of trunk groups Menu

1 - Display calls through any one gateway using the

trunk group(10)

2 - Display calls through all the gateways using the

trunk group(10)

0 - Previous menu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Thu

Jan

Jan

Jan

Jan

Jan

Jan

Jan

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Jan

Jan

Jan

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
5

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
5

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
5

13:49:50

13:49:50

13:49:50

13:49:50

13:49:50

13:49:50

13:49:50

13:49:50

13:49:50

13:49:50

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

Select 1 or 2, if 1 enter the SIP gateway number (0 to

999).

------------------------------------------Call ID

: [email protected]

Current state

: COMPLETED_STATE

From

: sip:32000@toto;user=phone

To

: sip:[email protected];user=phone

Gateway

: 151

Session Timer

: UPDATE method

------------------------------------------Number of
Calls in this Trunkgroup (10) : 1 (Active calls: 1)

-------------------------------------------

11 - SIP traces filters.

This functionality allows setting up to five filters on

SIP gateway calls. A filter is composed of the following

elements:

Filter string: String to search into the SIP calls

headers the user wants to trace.

From Field: If the field is set true, the user traces

 the SIP calls according to the

content of “From” header. In this case, if the SIP call

„From‟ header contains the filter

string defined for the filter, the SIP call will be

traced.

To Field: If the field is set true, the user traces the

SIP calls according to the content

of “To” header.

P_Asserted field: If the field is set true, the user

traces the SIP calls according to the

content of “P_Asserted” header.

Request-URI field: If the field is set true, the user

traces the SIP calls according to

the content of the Request URI.

Display conditions:

SIP call traces will be displayed if the SIP call

matches at least one of the five filters

of the array.

A SIP call matches to a filter if it fills one of the

conditions of the filter.

SIP traces filters menu

- Display the traces filters

- Add a traces filter

- Update a traces filter

- Remove a traces filter

- Remove all traces filters

- Previous menu

Ed. 07

63

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

1 - Display the traces filters.

--------------------------------------------------------
-----------------------| Nb | Filter

| From | To | P_Asserted | Request URI |

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
|-------------------------------------|------|------|---
---------|-------------|

| 1 | ...

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
| ... | ... |

...

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
...

|-------------------------------------|------|------|---
(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
---------|-------------|

| 2 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 3 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 4 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 5 | ...

| ... | ... |

...

...

--------------------------------------------------------
------------------------

2 - Add a traces filter.

String to filter ? (31 car. max) :

From field ? (y/n) :

To field ? (y/n) :

P_Asserted Field ? (y/n) :

Request URI field ? (y/n) :

Enter which information to filter (the filters are not

case sensitive), and manage on

each field to use it or not.

--------------------------------------------------------
-----------------------| Nb | Filter

| From | To | P_Asserted | Request URI |

|-------------------------------------|------|------|---
---------|-------------|

| 1 | alcatel-lucent.com

 | Yes | Yes |

Yes

Yes

|-------------------------------------|------|------|---
---------|-------------|

| 2 | genesys.com

| Yes | Yes |

Yes

Yes

|-------------------------------------|------|------|---
---------|-------------|

| 3 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 4 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 5 | ...

| ... | ... |

...

...

--------------------------------------------------------
------------------------

3 - Update a traces filter.

- Enter the filter number, in this case, only the filter

1 is managed.

From field ? (y/n) : y

To field ? (y/n) : y

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
P_Asserted Field ? (y/n) : n

Request URI field ? (y/n) : y

Ed. 07

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
64

TG0069

OmniPCX Enterprise

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
The filter string can not be modified, only on witch
field it is used.

--------------------------------------------------------
-----------------------| Nb | Filter

| From | To | P_Asserted | Request URI |

|-------------------------------------|------|------|---
---------|-------------|

| 1 | alcatel-lucent.com

| Yes | Yes |

No

Yes

|-------------------------------------|------|------|---
---------|-------------|

| 2 | genesys.com

| Yes | Yes |

Yes

Yes

|-------------------------------------|------|------|---
---------|-------------|

| 3 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 4 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 5 | ...

| ... | ... |

...

...

--------------------------------------------------------
------------------------

4 - Remove a traces filter.

--------------------------------------------------------
 -----------------------| Nb | Filter

| From | To | P_Asserted | Request URI |

|-------------------------------------|------|------|---
---------|-------------|

| 1 | alcatel-lucent.com

| Yes | Yes |

No

Yes

|-------------------------------------|------|------|---
---------|-------------|

| 2 | genesys.com

| Yes | Yes |

Yes

Yes

|-------------------------------------|------|------|---
---------|-------------|

| 3 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 4 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 5 | ...

| ... | ... |

...

...

--------------------------------------------------------
-----------------------Filter to remove (0:Previous
menu) :

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
-

Enter the filter number, only this one will be removed

(1 for instance).

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
--------------------------------------------------------
-----------------------| Nb | Filter

| From | To | P_Asserted | Request URI |

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
|-------------------------------------|------|------|---
---------|-------------|

| 1 | ...

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 2 | genesys.com

| Yes | Yes |

Yes

Yes

|-------------------------------------|------|------|---
---------|-------------|

| 3 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 4 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 5 | ...

| ... | ... |

...

...

--------------------------------------------------------
------------------------

Ed. 07

5 - Remove all traces filters.

- If you choose this option, all the filters will be

removed.

65

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Example: The traces must be done when alcatel-lucent.com

 is present on the “To” or the “From” field

and/or genesys.com on the “From” or the “P_Asserted”

fields .

The result is next:

--------------------------------------------------------
-----------------------| Nb | Filter

| From | To | P_Asserted | Request URI |

|-------------------------------------|------|------|---
---------|-------------|

| 1 | alcatel-lucent.com

| Yes | Yes |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 2 | genesys.com

| Yes | ... |

Yes

...

|-------------------------------------|------|------|---
---------|-------------|

| 3 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 4 | ...

| ... | ... |

...

...

|-------------------------------------|------|------|---
---------|-------------|

| 5 | ...

| ... | ... |

...

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
...

--------------------------------------------------------
(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
------------------------

When you will make a SIP trace (motortrace + traced),

the OXE will display the SIP exchanges and

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
information according to the filter management.

If you run the motortrace command and if a filter is

managed, the next message will be displayed:

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
motortrace (v5.2.0) verbosity = 00800004

The sipmotor traces level can not be changed

because some traces filters are set.

Please, remove them (with sipdump) before updating the

traces level.

Do not forget to remove all the filters after use.

12 - Display registred users.

Thu Jan 5 15:12:34 2012 --------------------------------

----------Thu Jan 5 15:12:34 2012

Detailed list of Registred users

Thu Jan 5 15:12:34 2012 --------------------------------

----------Thu Jan 5 15:12:34 2012

Thu Jan 5 15:12:34 2012

*************************************************

[CServRegistrar] Dump local registrar base

Address of record : 32003

contact : sip:[email protected]:46470, , 1611sec, 0.5

------------------------------------------------
Registrar statistics :

Number of users recorded : 1

Number of users having multiple contacts : 0

Number of contacts using UDP transport : 1

Number of contacts using TCP transport : 0

*************************************************

Thu Jan

5 15:12:34 2012 ----------------------------------------

---

Ed. 07

Comparing to the “sipregister” command, here there is

statistics about the Registrar.

66

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

11.5.7

Session Iniation Protcol (SIP)

Number of users recorded corresponds to the number of

SIP equipments registered

on the OXE Registrar.

Number of users having multiple contacts corresponds to

the SIP equipments with

multiple contacts, used in case of forking.

Number of contacts using UDP transport corresponds to

the number of contact

using UDP.

Number of contacts using TCP transport corresponds to

the number of contact using

TCP.

 sipextgw

This command is used with options:

sipextgw -l gives the external SIP gateways created and

their states.

===========================================================

| R E G I S T E R E D

S I P

E X T E R N A L

G A T E W A Y S |

===========================================================

IN SERVICE SIP external gateways list :

186

OUT OF SERVICE SIP external gateways list :

187

Here the external SIP gateway 186 is “in service” and

the external SIP gateway 187 is “out of service”.

sipextgw -g “external gateway number” gives the

configuration of this external SIP gateway.

===========================================================

S I P

E X T E R N A L

G A T E W A Y

Nb 186

===========================================================

Gateway Name

: SIMUL_SIP_ABCF

Gateway Type

: Standard type

State

: IN SERVICE

Belong to pool number

: -1

Use trunk group number

: 186 (ABC-F)

Remote domain

: 172.27.143.186

Port number

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
: 5060

Transport

: UDP

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
SRTP

: RTP only

Prack

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
: NO

Clir

: YES

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
SIP info enable

: NO

Authentication method

: NONE

SDP in 180 messages

: NO

Payload

: 97

Outgoing username

Outgoing password

: *****

Incoming username

Incoming password

: *****

Local domain name

Local user name

Realm name

Outbound proxy

Supervision timer

: 0

Registration timer

: 0

DNS type

: DNS A

Primary DNS IP address

: 000.000.000.000

Secondary DNS IP address : 000.000.000.000

PCS IP address

: 000.000.000.000

Retransmission number

of REGISTER/OPTIONS

: 2

Service route index

: -1

P-Asserted-ID

: FALSE

TrustedPAssIDHeader

: TRUE

Ed. 07

67

TrustedFromHeader

: FALSE

Diversion Info to

TG0069

 OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

This command is used to get a quick view of the

configuration given to this exteranl SIP gateway.

sipextgw -s “external gateway number” gives information

if the external SIP gateway is used

on a “Command table” (ARS) or/and a “Routing Number

Table”.

===========================================================

E X T E R N A L

G A T E W A Y

Nb 187

A R E A S

===========================================================

Found in ARS ==> dialling command table number : 187

Not found in ROUTING tables

Here the external SIP gateway 187 is used on the

“command table” 187.

===========================================================

E X T E R N A L

G A T E W A Y

Nb 186

A R E A S

===========================================================

Not found in ARS tables

Found in ROUTING table number : 12

Here the external SIP gateway 186 is used on the

“Routing table” 12.

11.5.8

sippool

this command is used to the external SIP gateways

associated to the same pool.

+-----------------------------------+

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
|

| pool Nb |

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
GW 1

GW 2

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
|

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
|

+-----------------------------------+

00

187 OOS | L 186

01

. . .

. . .

02

. . .

. . .

03

. . .

. . .

...

296 |

. . .

. . .

297 |

. . .

. . .

298 |

. . .

. . .

299 |

 . . .

. . .

+-----------------------------------+

Here the external SIP gateways 186 and 187 are in the
same pool, the pool number 0.

Ed. 07

"L"

shows the latter gateway used from the pool.

"OOS" means that the related gateway is OUT OF SERVICE.

68

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.5.9

sipdict

This command is used with options:

sipdict -l is used to list the sip users.

SIP DICTIONNARY, dim = 128, nb records = 16

+----------+----+---------------------------------------
-------+----+-----+------+------+------+-----+

| Ext. |

| mcdu

| i | url

|Type| Org | idx1 | idx2 | gw | Reg |

+----------+----+---------------------------------------
-------+----+-----+------+------+------+-----+

31020 | 0 |

31020@

oxe-ov | 3 |

1 |

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
12 |

0 | -- | -- |

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
31021 | 0 |

31021@

oxe-ov | 3 |

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
1 |

15 |

1 | -- | -- |

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
|

39002 | 0 |

39002@

oxeb-ov | 3 |

2 |

3 |

4 | -- | -- |

31853 | 1 |

31853@

opentouch-ov | 2 |

1 |

14 |

10 |

1 | No |

31022 | 0 |

31022@

oxe-ov | 3 |

1 |

1 |

11 | -- | -- |

31026 | 0 |

31026@

oxe-ov | 3 |

1 |

4 |

9 | -- | -- |

31040 | 0 |

31040@

oxe-ov | 2 |

1 |

10 |

5 | -- | -- |

31041 | 0 |

31041@

oxe-ov | 2 |

1 |

11 |

13 | -- | -- |

31028 | 0 |

31028@

oxe-ov | 3 |

1 |

9 |

8 | -- | -- |

31025 | 0 |

31025@

oxe-ov | 2 |

1 |

5 |

6 | -- | -- |

31023 | 0 |

31023@

oxe-ov | 3 |

1 |

13 |

7 | -- | -- |

31024 | 0 |

31024@

oxe-ov | 2 |

1 |

8 |

12 | -- | -- |

31852 | 0 |

31852@

oxe-ov | 1 |

1 |

0 |

3 | -- | -- |

31027 | 0 |

31027@

oxe-ov | 3 |

1 |

7 |

15 | -- | -- |

31854 | 0 |

31854@

oxe-ov | 3 |

1 |

6 |

14 | -- | -- |

31853 | 0 |

31853@

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
oxe-ov | 2 |

1 |

2 |

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
2 | -- | -- |

+----------+----+---------------------------------------
-------+----+-----+------+------+------+-----+

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
For each user directory number,the next information are
present:

- the “mcdu” corresponds to the directory number of the

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SIP user.

- the “i” is used to see if the SIP user is linked to an

external SIP gateway (0=no,

1=yes)

- the “dim” corresponds to the size of the dictionnary,

if the number of the SIP users

created is greater than 128, the OXE add one more 128 to
have 256, if the number

is greater than 256, the OXE add one more 128 to have
384, etc...the maximum is

128*80.

- the “url” corresponds to the SIP url known by the OXE.

the user 39002 is from another node (oxeb-ov).

- the “type” corresponds a SIP device or SIP extension:

1 is an external SIP voice mail.

2 is SIP device.

3 is SIP extension.

- the “org” corresponds to the origin node.

- the “idx1” and idx2” are assigned to the SIP users

during creation and used

internally.

- the “Ext.gw” is used in case of “Open Touch”

configuration, only for SIP device.

The user 31025 is using it, and it is know by the OXE by

31853@oxe-ov

and [email protected] on OXE.

172.27.143.186 is the “SIP Remote domain” managed on the

external SIP

gateway 186.

- the “Reg” is used to see if the user is registered on

the external SIP gateway.

Ed. 07

69

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

sipdict -i is used to list the sip users by using the

“idx1” (or “pos”).

SIP DICTIONNARY, dim = 128, nb records = 16

+------+----------+----+--------------------------------
----------------------+----+------+-----+

| Ext. |

 | pos | mcdu

| i | url par index

|Type| gw | Reg |

+------+----------+----+--------------------------------
----------------------+----+------+-----+

12 |

31852 | 0 |

31852@

oxe-ov | 1 | -- | -- |

15 |

31853 | 0 |

31853@

oxe-ov | 2 | -- | -- |

3 |

31853 | 1 |

31853@

172.27.143.186 | 2 | 186 | No |

14 |

31854 | 0 |

31854@

oxe-ov | 3 | -- | -- |

...

sipdict -v is used to list the sip users by using the

“idx2”.

sipdict -n “directory number of the SIP user” is used to

display the url associated.

(101)cpub_ov> sipdict -n 31027

Thu May 31 09:26:14 CEST 2012

URL = 31027@oxe-ov

sipdict -u “url of the SIP user” is used to display the

mcdu associated.

(101)cpub_ov> sipdict -u 31027 oxe-ov

Thu May 31 09:28:39 CEST 2012

31027@oxe-ov :

31027

Enter the url without the @ but just a space.

11.5.10

sipauth

This command is used with options:

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
o

sipauth -l is used to list the sip users.

SIP AUTHENTIFICATION, dim = 128, nb records = 13

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
+----------+--------------------------------------------
----------------+------+

| mcdu

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
| authentification

| idx1 |

+----------+--------------------------------------------
(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
----------------+------+

31020 |

31020 @

0000 |

2 |

31021 |

31021 @

0000 |

12 |

31853 |

31853 @

0000 |

1 |

31022 |

31022 @

0000 |

3 |

31026 |

31026 @

0000 |

9 |

31040 |

31040 @

0000 |

10 |

31041 |

31041 @

0000 |

8 |

31028 |

31028 @

0000 |

4 |

31025 |

31025 @

0000 |

11 |

31023 |

31023 @

0000 |

 7 |

31024 |

31024 @

0000 |

0 |

31027 |

31027 @

0000 |

6 |

31854 |

31854 @

0000 |

5 |

+----------+--------------------------------------------
----------------+------+

For each user directory number,the next information are

present:

Ed. 07

70

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

the “mcdu” correponds to the directory number of the SIP

user.

the “authentication” corresponds to the user login and

user pass for the

authentication, to managed on the SIP equipment if

needed.

the “idx1” is assigned to the SIP users during creation

and used internaly, same

than the one given by the “sipdict” command.

sipauth -i is used to list the sip users by using the

“idx1”.

SIP AUTHENTIFICATION, dim = 128, nb records = 13

+------+----------+-------------------------------------
-----------------------+

| pos | mcdu

| authentification

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
+------+----------+-------------------------------------
-----------------------+

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
2 |

31020 |

31020 @

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
0000 |

12 |

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
31021 |

31021 @

0000 |

1 |

31853 |

31853 @

0000 |

3 |

31022 |

31022 @

0000 |

9 |

31026 |

31026 @

0000 |

10 |

31040 |

31040 @

0000 |

8 |

31041 |

31041 @

0000 |

4 |

31028 |

31028 @

0000 |

11 |

31025 |

31025 @

0000 |

7 |

31023 |

31023 @

0000 |

0 |

31024 |

31024 @

0000 |

6 |

 31027 |

31027 @

0000 |

5 |

31854 |

31854 @

0000 |

+------+----------+-------------------------------------
-----------------------+

sipauth -n “directory number of the SIP user” is used to

display the user login and user pass.

(101)cpub_ov> sipauth -n 31027

Thu May 31 09:36:56 CEST 2012

LOGIN = 31027@0000

11.5.11

sipregister

This command is used with options:

sipregister, without option, display all the SIP and

SIPS users registered on registrar.

sipregister h

To get help menu.

*************************************************

Dump local registrar base

------------------------------------------------Address
of record : 31026

contact : sip:[email protected]:27836, udp, 502 s

------------------------------------------------Address
of record : 31022

contact : sip:[email protected], udp, 2867 s

------------------------------------------------Address
of record : 31853

contact : sip:[email protected], UDP, 319998256 s

------------------------------------------------Address
of record : 31023

contact : sip:[email protected]:1714, udp, 3300 s

------------------------------------------------Address
of record : 31027

contact : sip:[email protected], udp, 840 s

*************************************************

Ed. 07registred user number : 5

******

*************************************************

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
71

TG0069

OmniPCX Enterprise

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

For each address of record,the next information are

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present and given by the remote SIP equipment during

registration:

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the “contact” corresponds to the SIP address of the SIP
equipment with the IP

address to locate it.

the “upd” corresponds to the transport type, tcp can be

shown if it is used.

The “xx s” corresponds to the registration time left.

If no port number, the OXE will use the port 5060

sipregister l provides all the SIP users registered on

the registrar (option c is used for SIPS

users)

sipregister h

To get help menu.

*************************************************

Dump local registrar base

------------------------------------------------Address
of record : 31026

contact : sip:[email protected]:27836, udp, 502 s

------------------------------------------------Address
of record : 31022

contact : sip:[email protected], udp, 2867 s

------------------------------------------------Address
of record : 31853

contact : sip:[email protected], UDP, 319998256 s

------------------------------------------------Address
of record : 31023

contact : sip:[email protected]:1714, udp, 3300 s

------------------------------------------------Address
of record : 31027

contact : sip:[email protected], udp, 840 s

*************************************************

******

For each

registred

address

user

of record,the

number : 5next information are

*************************************************

registration:

11.5.12

present and given by the remote SIP equipment during

csipsets

This command is used with options:

csipsets with no option provides all the SIP extension

created on OXE.

+-----+--------+----------------+---------------+-----+

|Neqt |Number |Name

|IP address

|State|

 +-----+--------+----------------+---------------+-----+

|02054|31020

|MyIc_touch 172.2|

Unused| HS |

|02055|31027

|OT4135

| 172.27.143.184| ES |

|02058|31021

|RO31021

Unused| HS |

|02059|31022

|31022

| 172.27.141.206| HS |

|02061|31026

|31026

| 172.27.141.210| ES |

|02064|31028

|MyIC_phone

Unused| HS |

|02066|31023

|31023

Unused| HS |

|02068|31854

|31854

Unused| ES |

+-----+--------+----------------+---------------+-----+

|Number of SIP extensions: 00008

+-----------------------------------------------------+

For each user directory number,the next information are

present:

Ed. 07

72

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

the “Neqt” correponds to the equipment number of the SIP

extension given during its

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creation.

the “Number” corresponds to the directory number of the

SIP extension.

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
the “Name” corresponds the name of the SIP extension.

the “IP address” corresponds to the IP address of the

SIP equipment associated to

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
this SIP extension, if “Unused” is shown, that means
that no SIP equipment is

registered for this user.

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
the “State” corresponds to the status of the SIP
extension:

HS means that the user is Out Of Service.

ES means that the user is In Service.

The combination of the “IP address” and the “State”

gives you more information:

If the “IP address” is “Unused” and the “State” is ES:

the user is created, but no SIP equipment has been

registered for this user.

If the “IP address” is “Unused” and the “State” is HS:

the user has been already registered, but not anymore.

If the “IP address” is full with an IP address and the

“State” is HS:

the user is registered, but the user is Out Of Service,

this can be possible

due to the “keep alive” mechanism for SIP extension.

After registartion, the

SIP extension doesn‟t send or answer to the OPTION

messages.

If the “IP address” is full with an IP address and the

“State” is ES:

the user is registrered and In Service.

csipsets d “directory number” gives the information only

for this user.

(101)cpub_ov> csipsets d 31026

Mon Jun 4 14:08:56 CEST 2012

+-----+--------+----------------+---------------+-----+

|Neqt |Number |Name

|IP address

|State|

+-----+--------+----------------+---------------+-----+

|02061|31026

|31026

| 172.27.141.210| ES |

+-----+--------+----------------+---------------+-----+

csipsets n “neqt number” gives the information only for

this user.

(101)cpub_ov> csipsets n 2061

Mon Jun 4 14:09:54 CEST 2012

+-----+--------+----------------+---------------+-----+

|Neqt |Number |Name

|IP address

|State|

+-----+--------+----------------+---------------+-----+

|02061|31026

|31026

| 172.27.141.210| ES |

 +-----+--------+----------------+---------------+-----+

11.5.13

csipview com

Displays all the SIP extension calls.

No calls present, the display is next:

(101)cpub_ov> csipview com

Mon Jun 4 14:10:28 CEST 2012

+-----+--------+----------------+---------------+-------
-+

|Neqt |Number |Name

|IP address

|Activity|

+-----+--------+----------------+---------------+-------
-+

+-----+--------+----------------+---------------+-------
-+

Ed. 07

73 |

|Number

of SIP extensions in communication: 00000

+-------------------------------------------------------
-+

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Calls are present, the display is next:

(101)cpub_ov> csipview com

Mon Jun 4 14:13:41 CEST 2012

+-----+--------+----------------+---------------+-------
-+

|Neqt |Number |Name

|IP address

|Activity|

+-----+--------+----------------+---------------+-------
-+

|02061|31026

|31026

| 172.27.141.210|CH-CC

+-----+--------+----------------+---------------+-------
-+

|Number of SIP extensions in communication: 00001

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|

+-------------------------------------------------------
-+

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
For each user directory number,the next information are
present:

- the “Neqt” corresponds to the equipment number of the

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
SIP extension given during

its creation.

- the “Number” corresponds to the directory number of

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the SIP extension.

- the “Name” corresponds the name of the SIP extension.

- the “IP address” corresponds to the IP address of the

SIP equipment associated to

this SIP extension, if “Unused” is shown, that means

that no SIP equipment is

registered for this user.

- the “Activity” corresponds to the presence of a “Call

Control Half Com”. The “Call

Control Half Com”is in charge to interface the SIP world

to the OXE world.

11.5.14

csiprestart

This command is used with options:

csiprestart d “directory number” restarts the SIP

extension user:

(101)cpub_ov> csiprestart d 31026

Mon Jun

4 14:27:09 CEST 2012

csiprestart n “neqt number” restarts the SIP extension

user:

(101)cpub_ov> csiprestart n 2061

Mon Jun

4 14:27:09 CEST 2012

The option -f exist to force the restart if needed

11.5.15

sipextusers

(Only in R10.x for Open Touch).

This command is used with options:

sipextusers without option gives the list of the SIP

users associated to an Open Touch:

+---------+----------------------+------+----------+

| Number |Name

|Ext GW|Registered|

+---------+----------------------+------+----------+

| 60999

OXE_ADV_PROF|000001|

Yes|

| 60001

Dujardin Loulou|000001|

No|

| 60002

Lamy Chouchou|000001|

No|

 | 60050

Sy Omar|000001|

No|

+---------+----------------------+------+----------+

|Number of SIP USERS: 00004

+--------------------------------+

Ed. 07

74

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

sipextusers -d “directory number” of the SIP device

user:

+---------+----------------------+------+----------+

| Number |Name

|Ext GW|Registered|

+---------+----------------------+------+----------+

| 60001

Dujardin Loulou|000001|

No|

+---------+----------------------+------+----------+

For each user directory number,the next information are

present:

11.6

the “Number” corresponds to the directory number of the

SIP extension.

the “Name” corresponds the name of the SIP extension.

the “Ext GW” corresponds to the associated external SIP

gateway linked to this SIP

Device.

the “Registered” gives the information to know if the

SIP device is registered on

OXE side.

Link between SIPMOTOR traces and Call Handling traces

11.6.1

Call Handling / SIPMOTOR links implementation

CALL HANDLING

Local SIP

gateway

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
External SIP

gateway

CSIP (Call Control Half

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Com)

SIPMOTOR

The local SIP gateway “link” is used for the local SIP
(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
elements

- The SIP devices

- The external SIP Voice Mail

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The external SIP gateways “link” are used for the
connection between an external SIP equipment to the

OXE

- SIP carriers

- SIP applications (IVR, call center, etc...)

The Call Control Half Com “link” is used for the SIP
extension users (SEPLOS), it corresponds to the “CSIP”

function.

According to the declaration type of the SIP equipment

on the OXE, the behavior will be different on the

SIPMOTOR side, and also on the Call Handling side.

The exchanges between the SIPMOTOR to the Call Handling

are different according to this declaration.

Ed. 07

75

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.6.2

General view

When an issue appears in case of SIP equipment involved

on the communication, it is important to check if

the problem is from the SIPMOTOR or from the Call

Handling.

It is important to make the 2 traces simultaneously in

case of problem.

When a call is done, we can see on the motortrace the

exchange between the SIPMOTOR to the Call

handling.

Exchange from Call Handling to SIPMOTOR on SIPMOTOR

traces:

[display_ipc_in] ------------ Begin --------------.

[display_ipc_in] ------------- End ----------------

Exchange from SIPMOTOR to Call Handling on SIPMOTOR

traces:

[display_ipc_out] ------------ Begin --------------.

[display_ipc_out] ------------- End ----------------

Exchange from Call Handling to SIPMOTOR on Call Handling

traces:

+-------------------------------------------------------
-----+

| Message sent UA (neqt : XXXX-0) ----> SIP

Exchange from SIPMOTOR to Call Handling on Call Handling

traces:

+-------------------------------------------------------
 -----+

| Message received SIP ----> UA (neqt : XXXX)

Ed. 07

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Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.6.3

“neqt” link between SIPMOTOR and Call Handling traces

When traces are done on OXE to find the cause of the

issue, it is important to link the call on the SIPMOTOR

trace and the Call Hanling trace, for this check the
“neqt” number used (the neqt is 2250 in the next

examples)

On SIPMOTOR traces:

Mon

Mon

Mon

Mon

Mon

May

May

May

May

May

28

28

28

28

28

14:22:38

14:22:38

14:22:38

14:22:38

14:22:38

2012

2012

2012

2012

2012

For incoming call, the neqt is seen before the

“display_ipc_out” message:

[CMotorCallManager::insertCallwithEqt] CMotorCall 2250

inserted.

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11f7[sendLgEvtSipCreate] Event sent on eqt : 2250

[display_ipc_out] ------------ Begin --------------Id :

-1

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INVITE

For outgoing call, the neqt is given on the

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
“display_ipc_in” message from the Call

handling

Mon May 28 14:27:48 2012 [display_ipc_in] ------------

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Begin --------------Mon May 28 14:27:48 2012 neqt : 2250
Id : -1

Mon May 28 14:27:48 2012 INVITE

On Call Handling traces:

(215701:000005)

(215701:000006)

(215701:000007)

(215701:000008)

(215701:000009)

(215701:000010)

SIP : message INVITE arrive sur le neqt : 2250.

init_data_network

init_data_network FIN

SIP : ctrl_sip evt : 10752.

+-------------------------------------------------------
-----+

| Message received SIP ----> UA (neqt : 2250)

(222651:000188)

(222651:000189)

(222651:000190)

(222651:000191)

For incoming call, the neqt is seen with this message:

For outgoing call, the neqt is seen with this message:

SIP : [send_to_motor] ipcSend resultat : 0 sur eqt :

2250

SIP : [ipc_send] envoi du message : 10752.

+-------------------------------------------------------
-----+

| Message sent UA (neqt : 2250-0) ----> SIP

For traces analyses, follow all the exhanges using this

neqt, it is not possible to get more than one active
call

using this “neqt”. When the call is released, this

“neqt” is freed for another call.

The “neqt” number can correspond to:

A SIP extension, the same everytime.

A time slot of the SIP Trunk Group used on the local SIP
gateway for SIP

device user, different according to which time slot is

used.

A time slot of the SIP Trunk Group used on the local SIP
gateway for SIP

external Voice Mail, different according to which time

slot is used.

A time slot of the SIP Trunk Group used for the external
SIP gateway,

different according to which time slot is used.

Ed. 07

77

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 OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.7

Information on the SIPMOTOR traces

On the SIPMOTOR traces, information are between “[...]”.

These information are important to understand the

information after it and to troubleshoot the issue.

Examples:

[CCall::receiveRequest] INVITE: The SIPMOTOR has

received a SIP request and

the request is an INVITE.

[CTransaction::changeState]: The SIPMOTOR has changed

the state of a

transaction.

[getFromHeader]: the SIPMOTOR gets the information from

the FROM header in

case of SIP incoming call.

[isDomainFromGwExt]: the SIPMOTOR checks if the

information from the domain

part of the FROM corresponds to an external SIP gateway.

The information “event” and “message” are in relation

with the direction of the call and the SIP message:

“event” is for the Call Handling.

“message” is for the SIPMOTOR.

The information between the [...] are more or less

understandable, they can help to find the root cause of
the

issue.

11.8

Follow a call on the SIPMOTOR trace

For SIP point of view, the call can be followed by the

Call-ID, but on the SIPMOTOR, there are information

for calls distinctions

We have the “neqt” number, it is used to link the

SIPMOTOR and Call Handling traces

The Session reference is used to follow the call.

Mon May 28 15:21:04 2012

...

Mon May 28 15:21:04 2012

...

Mon May 28 15:21:04 2012

ov.alcatel.fr;user=phone

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
...

Mon May 28 15:21:04 2012

...

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Mon May 28 15:21:04 2012

Ed. 07

On this example, the Session reference is “1173”

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
1173[CMotorCall::sipUriType] sip Uri.

1173[CMotorCall::getUserType] seplos station crypto=0.

1173[CMotorCall::emitInviteMessage]

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
1173[CMotorCall::inviteBuildContact]

To: "Xlite PC" sip:31023@oxeContact: sip:31004@oxe-

ov.alcatel.fr

1173 [CCall::makeGenericRequest] INVITE

To find this Session reference for an outgoing call,

search for

“[CMotorCall::sipUriType] sip Uri.” before the INVITE

sent to the remote SIP

equipment.

To find this Session reference for an incoming call,

search for

“[CCall::receiveRequest] INVITE” after the INVITE

received from the remote SIP

equipment.

78

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The transation reference, this value can be used to

follow the transaction status evolution and to get

information about this transaction

Mon

...

Mon

...

Mon

Mon

...

Mon

...

Mon

On this example, the transaction reference is “21be”

May 28 15:21:04 2012 21be [CTransaction::changeState]

STATE CHANGED TO INITIAL

May 28 15:21:04 2012 21be [CTransaction::changeState]

STATE CHANGED TO CALLING

May 28 15:21:04 2012 21be [CTransaction::startTimer]

Timer A is started (delay = 500 ms)

May 28 15:21:04 2012 21be [CTransaction::startTimer]

Timer B is started (delay = 4000 ms)

May 28 15:21:04 2012 21be [CTransaction::changeState]

STATE CHANGED TO PROCEEDING

May 28 15:21:08 2012 21be [CTransaction::changeState]

STATE CHANGED TO TERMINATED

To find this transaction reference for an outgoing call,

search for “STATE

 CHANGED TO INITIAL” before the INVITE sent to the remote

SIP equipment.

To find this transaction reference for an incoming call,

search for “STATE

CHANGED TO INITIAL” after the INVITE received from the

remote SIP

equipment.

For one transaction, there is a pair of reference, a

“clone” reference associated to

the main one, if the main one is 21be, the second

reference is 21bf associated with

the 200ok receive or sent. This reference is seen with

this message after the 200ok.

Mon May 28 15:21:08 2012 21bf

[CTransaction::CTransaction] Transaction is cloned in 4
state

The dialog reference, this value can be used to follow

the dialog evolution and to get information

about this dialog

- On this example, the dialog reference is “158a”

Mon May 28

Mon May 28

Mon May 28

...

Mon May 28

= Initial,

...

Mon May 28

Mon May 28

15:21:04 2012 158a [CDialog::createRequest]

15:21:04 2012 158a [CDialog::buildServicesForAllRequest]

15:21:04 2012 158a [CDialog::createInviteRequest]

15:21:04 2012 158a [CDialog::onTransactionState(pTrans =

21be, previousState = Terminated, currentState

reason = None]

15:21:08 2012 158a [CDialog::receiveResponse]

15:21:08 2012 158a [CDialog::receiveResponse] create a

CONFIRMED dialog

To find this dialog reference for an outgoing call,

search for

“CDialog::createRequest” before the INVITE sent to the

remote SIP equipment.

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
-

To find this dialog reference for an incoming call,

search for

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
“CDialog::receiveRequest” after the INVITE received from
the remote SIP

equipment.

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
-

For one dialog, there is a pair of reference, a “clone”

reference associated to the

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main one, if the main one is 158a, the second reference
is 158b associated with the

200ok receive or sent. This reference is seen with this

message after the 200ok.

Mon May 28 15:21:08 2012 158b [CDialog::CDialog] look

for the transaction #0, transaction key =

z9hG4bKca60f1097ab026913ca3bf56995162be

Ed. 07

79

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

This Information links the transaction to the dialog.

Mon May 28 15:21:04 2012 158a

[CDialog::onTransactionState(pTrans = 21be,
previousState = Terminated, currentState

= Initial, reason = None]

For the dialog, the transaction reference is linked. The

dialog “158a” is linked to the

transaction “21be”.

There is the same link for the “clone” references.

Mon May 28 15:21:08 2012 158b

[CDialog::onTransactionState(pTrans = 21bf,
previousState = Proceeding, currentState

= Completed, reason = Final resp reception]

The SIPMOTOR is using references for INVITE treatment:

The Session reference, this one is unique for the

complete call (from INVITE to the 200ok of the

BYE)

The Dialog references, 2 references are used:

o The main one is created when the INVITE is sent or

received

o The clone one, used to change the dialog state

according to the transactions used for a new

event on the call (put on hold, transfer, etc...)

The Transaction references, the number of references

depends of the call events (put on hold,

transfer, etc...)

o The main one is created when the INVITE is sent or

received

o The other ones are created if an event is coming for

the dialog associated (ACK, BYE,

REINVITE, REFER, etc...)

A permanent link is done between the Dialog (main and

clone) to the Transactions (main and clones), here

an example for an incoming call with 2 REINVITEs and a

BYE at the end:

 UAC

. . . . .

UAS

(SIP set)

(Proxy)

|(1) INVITE

|-------------------->|

|(2) 100 Trying

|<--------------------|

|(3) 180 Ringing

|<--------------------|

|(4) 200 OK

|<--------------------|

|(5) ACK

|-------------------->|

|(6) INVITE

|-------------------->|

(1) Assignation a reference to the session, dialog and

transaction

(4) Creation of the clone dialog and the first clone

transaction,

associated to the clone dialog

(5) First clone transaction terminated

(6) Creation of the second clone transaction for the

first REINVITE,

associated to the clone dialog

(8) Second clone transaction terminated

Ed. 07

(9) Creation

of the third clone transaction for the second

80

TG0069

REINIVTE, associated to the clone dialog

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

|(7) 200 OK

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
|

|<--------------------|

|(8) ACK

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
|

|-------------------->|

|(9) INVITE

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
|

|-------------------->|

|(10) 200 OK

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
|

|<--------------------|

|(11) ACK

|-------------------->|

|(12) BYE

|-------------------->|

|(13) 200 OK

|<--------------------|

11.9

Traces analyses

11.9.1

Incoming SIP call using a SIP Trunk Group: SIPMOTOR

point of view

Here an example of incoming call from a SIP device to an

IPtouch.

Mon May 28 16:41:57 2012 RECEIVE MESSAGE FROM NETWORK

(135.118.226.39:25648 [UDP])

----------------------utf8----------------------INVITE
sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 135.118.226.39:25648;branch=z9hG4bK-

d87543-46534e582323f252-1--d87543-;rport

Max-Forwards: 70

Contact: <sip:[email protected]:25648>

To: "31004"<sip:[email protected]>

From: "PC_sip_device"<sip:31024@oxe-
ov.alcatel.fr>;tag=f6448c0c

Call-ID: ZWEwMGI4YjUxNjMyOWRlZmEyNWEzYThmNzI4NDUzMGM.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,

MESSAGE, SUBSCRIBE, INFO

Content-Type: application/sdp

User-Agent: Sip Phone

Content-Length: 315

v=0

o=- 3 2 IN IP4 135.118.226.39

s=Sip_Phone

c=IN IP4 135.118.226.39

t=0 0

m=audio 7888 RTP/AVP 8 18 101

a=fmtp:18 annexb=no

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

Ed. 07

a=sendrecv

a=x-rtp-session-id:A56A9738C0BC4CEF8087E10840231621

-------------------------------------------------

81

 TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The information “RECEIVE MESSAGE FROM NETWORK

(135.118.226.39:25648 [UDP])” is important to

know that the call is an incoming one from the SIP

equipment 135.118.226.39 in UDP.

The SIPMOTOR checks the Call-Id to know if this INVITE

is an INVITE or a REINVITE.

Mon May 28 16:41:57 2012 1153 [CCall::getDialog]

Confirmed Dialog is not found (ID = ;f6448c0c)

Mon May 28 16:41:57 2012 1153 [CCall::getDialog] Initial

Dialog Server not found

Here, it is an INVITE, because the dialog is not found.

The transaction and the dialog are put in place.

Mon May 28 16:41:57 2012 21a5

[CTransaction::changeState] STATE CHANGED TO INITIAL

...

Mon May 28 16:41:57 2012 156c

[CDialog::onTransactionState(pTrans = 21a5,
previousState = Terminated, currentState

= Initial, reason = None]

Here, the transaction reference is “21a5” and the dialog

reference is “156c”.

The transaction status is changed, because the dialog is

initiated.

Mon May 28 16:41:57 2012 21a5

[CTransaction::changeState] STATE CHANGED TO PROCEEDING

Mon May 28 16:41:57 2012 21a5

[CTransaction::changeState] notifying the parent dialog

When a transaction is linked to a dialog, the

transaction changed from INITIAL to PROCEEDING.

The SIPMOTOR generates the 100 Trying.

Mon May 28 16:41:57 2012 SEND MESSAGE TO NETWORK

(135.118.226.39:25648 [UDP]) (BUFF LEN = 346)

----------------------utf8----------------------SIP/2.0
100 Trying

To: "31004" <sip:[email protected]>

From: "PC_sip_device" <sip:31024@oxe-

ov.alcatel.fr>;tag=f6448c0c

Call-ID: ZWEwMGI4YjUxNjMyOWRlZmEyNWEzYThmNzI4NDUzMGM.

CSeq: 1 INVITE

Via: SIP/2.0/UDP
135.118.226.39:25648;received=135.118.226.39;branch=z9hG4bK
(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
d87543-46534e582323f252-1--d87543;rport=25648

Content-Length: 0

-------------------------------------------------

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
The SIPMOTOR checks the Session Timer for the call.

Mon May 28 16:41:57

(Server, UA)

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Mon May 28 16:41:57

reception)

Mon May 28 16:41:57

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Mon May 28 16:41:57

2012 [CSessionTimerContext::CSessionTimerContext] New

CSessionTimerContext from request

2012 [CSessionTimerContext::updateAfterRefreshReception]
Update CSessionTimerContext (refresh

2012 [CSessionTimerContext::updateSessionExpires]
Session-Expires updated : 0

2012 [CSessionTimerContext::setRefreshMethod] Allow

refreshMethod=INVITE

In this case, the SIP equipment doesn‟t send “Session

timer” information because the value is 0 (updated :

0).

Ed. 07

82

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The SIPMOTOR makes the link between the dialog,

transaction, the branch and the Cseq number.

Mon May 28 16:41:57 2012 156c [CDialog::addTransaction]

added transaction 21a5 with branch z9hG4bK-
d8754346534e582323f252-1--d87543-, with CSeq 1

The “branch” is a parameter added to the “via” to

identify it. Regarding rfc3261, all the branch

values must start by “z9hG4bK”.

The CSeq is used to indentify and to order a

transaction, it consists of a sequence number and a
method.

The SIPMOTOR checks for which OXE equipment the call is

from.

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

May

May

May

May

May

May

May

 May

May

May

May

May

28

28

28

28

28

28

28

28

28

28

28

28

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

[isDomainFromGwExt] Host from request is :

172.27.142.53.

[isDomainFromGwExt] User from request is : 31024

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
[domain not from an External Gateway.

1153[CMotorCall::setFilterUsedMode] To be traced = 0

1153[CMotorCall::initOfUserType] values are reseted

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
[getFromHeader] displayName="PC_sip_device".

[getFromHeader] [email protected].

[getFromHeader] clirPresent=0.

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
[isAddrInDico] user=31024 host=oxe-ov.alcatel.fr

[isUserInDico] [email protected]

[isUserInDico] found in the dictionnary.

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
[isAddrInDico] sip device station OK

The SIPMOTOR checks first if the domain part from the

PAI, and of the FROM if no PAI,

to see if the call is for an external SIP gateway.

Here, we can see that the call is from a SIP Device.

The SIPMOTOR checks for who the call is done .

Mon

Mon

Mon

Mon

May

May

May

May

28

28

28

28

16:41:57

16:41:57

16:41:57

16:41:57

2012

2012

2012

2012

[isAddrInDico] user=31004 host=oxe-ov.alcatel.fr

[isUserInDico] [email protected]

isUserInDico] NOT found in the dictionnary.

[isAddrInDico] other sip user

Here the call is for an “other sip user”, that means the
call is for a non SIP user, corresponding to a legacy

set (IPtouch).

The SIPMOTOR checks the number of licenses available.

Mon May 28 16:41:57 2012

1153[CMotorCall::methodInviteReceived] nb available
licenses=25

Here the number of licenses is 25, that means, 25 calls

are possible on SIP using a SIP Trunk Group or

SEPLOS users.

The SIPMOTOR checks if the IP address recieved is

managed on an IP domain.

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

 ...

Mon

May

May

May

May

May

May

May

May

28

28

28

28

28

28

28

28

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

2012

2012

2012

2012

2012

2012

2012

2012

May 28 16:41:57 2012

The recevied host 135.118.226.39

Trying to find the ip address in domain list

The entry dom : 141 add_type=1

The entry dom ip low :172.27.141.165

The entry ipaddress from low :135.118.226.39

The entry compare :1

The entry compare 2 :0

iplink_is_good_range_for_reg

The user domain is

142

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Here, the IP address of the SIP equipment corresponds to
the IP domain 142.

If the IP address doesn‟t match an IP domain, the

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
SIPMOTOR returns:

Ed. 07

83

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
TROUBLESHOOTING GUIDE No.0069

Mon May 28 16:41:57 2012

The user is ipadd

not in any Domain range return state as -1

The SIPMOTOR checks the SDP received on the INVITE.

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

May

May

May

May

May

May

May

May

May

May

May

May

May

May

May

May

May

May

May

28

28

28

28

28

28

28

 28

28

28

28

28

28

28

28

28

28

28

28

16:41:57

16:41:57

16:41:57

16:41:57

16:41:57

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2012

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[checkSdpValidity] Media 0 type 1 contains 3 formats.

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
[checkSdpValidity] Format : 8.

1153[CMotorCall::isCryptoAuthorized] user crypto=0.

[convertSdpIntoTsdp] No Direction in the session part.

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
[convertSdpIntoTsdp] Check the direction in Session part
- result:0.

[convertSdpIntoTsdp] media AUDIO detected (previous

crypto=0).

[convertAudioMedia] The audio media contains 3

format(s).

[convertAudioMedia] Format 0 is 8.

[convertAudioMedia] Format 1 is 18.

[convertAudioMedia] Format 2 is 101.

[convertAudioMedia] 101.

[convertAudioMedia] Format is DTMF:101.

[convertAudioMedia] Direction is sendrecv.

[convertAudioMedia] Connection address retrieved in sdp:

135.118.226.39.

[convertIPStrIntoTuipv] 135.118.226.39 => 135.118.226.39

[display_sdp] address =135.118.226.39

[display_sdp] direction=0.

[convertSdpIntoTsdp] only one media taken into account

xxx crypto_index=0 clear media=1

[convertSdpIntoTsdp] crypto_index=0 clear media=1.

The SDP contains in this SDP three formats of medias (8,

18 and 101), the direction is “sendrecv” meaning

in both direction and the IP address of connection is

135.118.226.39.

The message to Call Handling is prepared and sent to it.

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(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
16:41:57

16:41:57

16:41:57

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
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Ed. 07

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1153[sendLgEvtSipCreate] Event sent on eqt : 2250

[display_ipc_out] ------------ Begin --------------Id :

-1

INVITE

REQUEST URI : <> [email protected]:5060 ;

user=name

FROM : <PC_sip_device> [email protected]:5060 ;

user=name

TO : <"31004"> [email protected]:5060 ; user=name

CAC : 0

CAC ADDRESS :

CAC-CSBU info : UNKNOWN

CLIR : 0

 Prack Required : 0

Allow Update : 0

SDP :

ADDRESS : 135.118.226.39 :7888

ALGOS :

PCMA

G729

101

DIRECTION : SEND & RECEIVE

crypto index : 0

N_GW_EXT : -1

[display_ipc_out] ------------- End ----------------

84

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The call is sent to the Call handling on neqt 2250,

regarding the type of SIP equipment detected by the

SIPMOTOR, some information are added or not on this

message.

All the information about this call are sent to the

Stand-By CPU.

Mon May 28 16:41:57 2012 SendToSipgwCpuSec: Message sent

to the STAND-BY CPU

Mon May 28 16:41:57 2012 [receiveInviteMessage] send

RemoteSdp to the StandBy.

Mon May 28 16:41:57 2012 SendToSipgwCpuSec: Message sent

to the STAND-BY CPU

The information are sent to the Stand-By, like this, in

case of bascul the SIP call will not be lost and known

on the new main CPU

The Call handling sends back an answer for this INVITE.

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[display_ipc_in] ------------ Begin --------------neqt :

2250 Id : -1

INFORMATIONAL

xx :

80

RELATIVE REQUEST : INVITE

[display_ipc_in] ------------- End ----------------

A “180 Ringing” is sent to the SIPMOTOR without SDP

The Call handling sends back an answer for this INVITE.

Mon May 28 16:41:57 2012 SEND MESSAGE TO NETWORK

(135.118.226.39:25648 [UDP]) (BUFF LEN = 547)

----------------------utf8----------------------SIP/2.0
180 Ringing

Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER,

SUBSCRIBE, OPTIONS, UPDATE

Contact: sip:oxe-ov.alcatel.fr

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

To: "31004" <sip:31004@oxe-

ov.alcatel.fr>;tag=15654dedb5658c165fbba7b0026e6ae9

From: "PC_sip_device" <sip:31024@oxe-

ov.alcatel.fr>;tag=f6448c0c

Call-ID: ZWEwMGI4YjUxNjMyOWRlZmEyNWEzYThmNzI4NDUzMGM.

CSeq: 1 INVITE

Via: SIP/2.0/UDP
135.118.226.39:25648;received=135.118.226.39;branch=z9hG4bK
d87543-46534e582323f252-1--d87543;rport=25648

Content-Length: 0

-------------------------------------------------

A “180 Ringing” is sent to the SIPMOTOR without SDP

For each SIP call event, a message is send to the Stand-

By CPU.

Mon May 28 16:41:57 2012 [receiveInformationalEvent]

UpdateContext send on the StandBy.

The Call handling sends a new answer for this INVITE.

Ed. 07

85

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

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Session Iniation Protcol (SIP)

[display_ipc_in] ------------ Begin --------------neqt :

2250 Id : -1

SUCCESSFUL

xx :

RELATIVE REQUEST : INVITE

CLIR : 0

COLP : 1

CAC-CSBU info : UNKNOWN

SDP :

ADDRESS : 172.27.142.64 :32514

ALGOS :

G729

101

DIRECTION : SEND & RECEIVE

crypto index : 0

[display_ipc_in] ------------- End ----------------

A “200 ok” is sent to the SIPMOTOR with SDP

The SIPMOTOR checks if the SDP given is compatible with

the SDP received in the INVITE.

Mon May 28 16:41:58 2012

1153[CMotorCall::makeResponseSdp] Audio media.

Mon May 28 16:41:58 2012

1153[CMotorCall::appendAudioAttributToMedia] Direction:
0.

 Mon May 28 16:41:58 2012

1153[CMotorCall::appendAudioAttributToMedia] format 101

Mon May 28 16:41:58 2012

1153[CMotorCall::makeResponseSdp]
fromSdp.getMediaDesciprionCount :1

Mon May 28 16:41:58 2012 [sameCodec] accepted Format :

18.

Mon May 28 16:41:58 2012 [sameCodec] requested Format :

8.

Mon May 28 16:41:58 2012 [sameCodec] requested Format :

18.

Mon May 28 16:41:58 2012 [sameCodec] same Format.

Mon May 28 16:41:58 2012 1153[CMotorCall::mediaAccepted]

Media accepted: m=audio 32514 RTP/AVP 18 101

a=sendrecv

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=ptime:20

a=maxptime:40

a=rtpmap:101 telephone-event/8000

The codecs from the INVITE were 8 and 18, on the answer
we have 18, in that case the call is accepted by

SIPMOTOR for SDP point of view.

The SIPMOTOR is changing the status of the dialog.

Mon May 28 16:41:58 2012 156c [CDialog::createResponse]

create a CONFIRMED dialog

Due to this, the dialog reference and transaction

reference are changed (internal SIPMOTOR functionning).

Mon May 28 16:41:58 2012 156d [CDialog::CDialog] look

for the transaction #0, transaction key = z9hG4bK-
d8754346534e582323f252-1--d87543Mon May 28 16:41:58 2012
156d [CDialog::CDialog] copy the transaction #0,
transaction key = z9hG4bK-d8754346534e582323f252-1--
d87543Mon May 28 16:41:58 2012 21a6
[CTransaction::CTransaction] Transaction is cloned in 4
state

The dialog reference is changed form “156c” to “156d”.

The transaction reference is changed from “21a5” to

“21a6”.

The SIPMOTOR is changing the status of the dialog.

1338216118 -> Mon May 28 16:41:58 2012 SEND MESSAGE TO

NETWORK (135.118.226.39:25648 [UDP]) (BUFF LEN = 974)

----------------------utf8----------------------SIP/2.0
200 OK

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER,
SUBSCRIBE, OPTIONS, UPDATE

Contact: sip:oxe-ov.alcatel.fr

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Supported: replaces,timer,path,100rel

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

Session-Expires: 1800;refresher=uas

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
P-Asserted-Identity: "IPtouch 172.27.1" <sip:31004@oxe-
ov.alcatel.fr;user=phone>

Ed. 07 application/sdp

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
86

TG0069

Content-Type:

To: "31004" <sip:31004@oxe-

ov.alcatel.fr>;tag=15654dedb5658c165fbba7b0026e6ae9

From: "PC_sip_device" <sip:31024@oxe-

ov.alcatel.fr>;tag=f6448c0c

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The 200ok sent to the remote SIP equipment is generated

with information from the INVITE received and

from the 200ok answer from the Call Handling.

The SIPMOTOR is changing the status of the transaction.

Mon May 28 16:41:58 2012 21a6

[CTransaction::changeState] STATE CHANGED TO COMPLETED

The retransmission timers are started.

Mon May 28 16:41:58 2012 21a6 [CTransaction::startTimer]

Timer G is started (delay = 500 ms)

Mon May 28 16:41:58 2012 21a6 [CTransaction::startTimer]

Timer H is started (delay = 32000 ms)

The SIPMOTOR receives a ACK for the 200ok.

Mon May 28 16:41:59 2012 RECEIVE MESSAGE FROM NETWORK

(135.118.226.39:25648 [UDP])

----------------------utf8----------------------ACK
sip:oxe-ov.alcatel.fr SIP/2.0

Via: SIP/2.0/UDP 135.118.226.39:25648;branch=z9hG4bK-

d87543-b00f692e5d3a246e-1--d87543-;rport

Max-Forwards: 70

Contact: <sip:[email protected]:25648>

To: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=15654dedb5658c165fbba7b0026e6ae9

From: "PC_sip_device"<sip:31024@oxe-
ov.alcatel.fr>;tag=f6448c0c

Call-ID: ZWEwMGI4YjUxNjMyOWRlZmEyNWEzYThmNzI4NDUzMGM.

CSeq: 1 ACK

User-Agent: Sip Phone

Content-Length: 0

-------------------------------------------------

The SIPMOTOR is changing the status of the transaction.

Mon May 28 16:41:59 2012 21a6

[CTransaction::changeState] STATE CHANGED TO TERMINATED

The retransmission timers are freed.

Mon May 28 16:41:59 2012 21a6

[CTransaction::freeTimerToken] Timer G is freed

Mon May 28 16:41:59 2012 21a6

[CTransaction::freeTimerToken] Timer H is freed

The SIPMOTOR is changing the status of the dialog.

Ed. 07

87

 TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Mon May 28 16:41:59 2012 156d

[CDialog::receiveAckRequest] the INVITE request is
terminated

The ACK is sent to the Call Handling.

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[display_ipc_out] ------------ Begin --------------Id :

-1

ACK

[display_ipc_out] ------------- End ----------------

After call establishment, the call can be released by

the OXE or by the remote SIP equipment.

Call released by the Call Handling:

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The BYE is sent from the Call Handling.

2012

2012

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2012

2012

[display_ipc_in] ------------ Begin --------------neqt :

2250 Id : -1

BYE

[display_ipc_in] ------------- End ----------------

Creation of a new transaction for the BYE.

Mon May 28 16:42:00 2012 21a7

[CTransaction::changeState] STATE CHANGED TO INITIAL

The BYE is a new transaction for a SIP call, in that

case, the transaction reference it is “21a7”, and the
status

is “INITIAL”.

The BYE is sent to the remote SIP equipment.

Mon May 28 16:42:00 2012 SEND MESSAGE TO NETWORK

(135.118.226.39:25648 [UDP]) (BUFF LEN = 454)

----------------------utf8----------------------BYE
sip:[email protected]:25648 SIP/2.0

Supported: replaces,timer,path,100rel

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

To: sip:[email protected];tag=f6448c0c

From: "31004" <sip:31004@oxe-

ov.alcatel.fr>;tag=15654dedb5658c165fbba7b0026e6ae9

Call-ID: ZWEwMGI4YjUxNjMyOWRlZmEyNWEzYThmNzI4NDUzMGM.

CSeq: 1948273321 BYE

Via: SIP/2.0/UDP
172.27.142.53;branch=z9hG4bK9f0b6b39121b23d361a5f6a8101aaa9

Max-Forwards: 70

Content-Length: 0

-------------------------------------------------

The SIPMOTOR changes the transaction state.

Mon May 28 16:42:00 2012 21a7

[CTransaction::changeState] STATE CHANGED TO TRYING

The retransmission timers are started.

Mon May 28 16:42:00 2012 21a7 [CTransaction::startTimer]

Timer E is started (delay = 500 ms)

Mon May 28 16:42:00 2012 21a7 [CTransaction::startTimer]

Timer F is started (delay = 16000 ms)

Ed. 07

The 200ok of the BYE request is received from the remote

SIP equipment.

88

TG0069

OmniPCX Enterprise

 Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Mon May 28 16:42:00 2012 RECEIVE MESSAGE FROM NETWORK

(135.118.226.39:25648 [UDP])

----------------------utf8----------------------SIP/2.0
200 OK

Via: SIP/2.0/UDP
172.27.142.53;branch=z9hG4bK9f0b6b39121b23d361a5f6a8101aaa9

Contact: <sip:[email protected]:25648>

To: <sip:[email protected]>;tag=f6448c0c

From: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=15654dedb5658c165fbba7b0026e6ae9

Call-ID: ZWEwMGI4YjUxNjMyOWRlZmEyNWEzYThmNzI4NDUzMGM.

CSeq: 1948273321 BYE

User-Agent: Sip Phone

Content-Length: 0

-------------------------------------------------

The SIPMOTOR changes this transaction state.

Mon May 28 16:42:00 2012 21a7

[CTransaction::changeState] STATE CHANGED TO COMPLETED

The retransmission timers are freed.

Mon May 28 16:42:00 2012 21a7

[CTransaction::freeTimerToken] Timer E is freed

Mon May 28 16:42:00 2012 21a7

[CTransaction::freeTimerToken] Timer F is freed

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The 200ok of the BYE request is sent to the Call

Handling.

16:42:00

16:42:00

16:42:00

16:42:00

The Call Handling sent a message to the SIPMOTOR to

release the “neqt” associated to

this SIP call

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16:42:00

16:42:00

16:42:00

16:42:00

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2012

[display_ipc_in] ------------ Begin --------------neqt :

2250 Id : -1

SIP EQT RELEASED

[display_ipc_in] ------------- End ----------------

The SIPMOTOR acknowledge the release of the “neqt”

Mon

Mon

Mon

Mon

[display_ipc_out] ------------ Begin --------------Id :

-1

SUCCESSFUL

xx :

RELATIVE REQUEST : BYE

CAC-CSBU info : UNKNOWN

CLIR : 0

COLP : 0

[display_ipc_out] ------------- End ----------------

[display_ipc_out] ------------ Begin --------------Id :

-1

SIP_EQT_RELEASE_ACK

[display_ipc_out] ------------- End ----------------

The SIPMOTOR kills the SIP call

Mon May 28 16:42:00 2012

[CMotorCallManager::onIncomingEvent] killSession.

Mon May 28 16:42:00 2012 1153 [CCall::killSession]

The SIPMOTOR changes the state of the transactions

Mon May 28 16:42:00 2012 21a5

[CTransaction::changeState] STATE CHANGED TO TERMINATED

...

Mon May 28 16:42:00 2012 21a7

[CTransaction::changeState] STATE CHANGED TO TERMINATED

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Ed. 07

89

TG0069

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Call released by the remote SIP equipment:

The BYE is received from the remote SIP equipment.

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Mon May 28 16:42:00 2012 RECEIVE MESSAGE FROM NETWORK
(135.118.226.39:25648 [UDP])

----------------------utf8----------------------BYE
sip:oxe-ov.alcatel.fr SIP/2.0

Via: SIP/2.0/UDP 135.118.226.39:25648;branch=z9hG4bK-

d87543-cf501c2f3311d050-1--d87543-;rport

Max-Forwards: 70

Contact: <sip:[email protected]:25648>

To: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=ba904e80f620e0f32593273ec97e818d

From: "PC_sip_device"<sip:31024@oxe-
ov.alcatel.fr>;tag=b05ced13

Call-ID: NTEwZjI0M2VjZGY1YzExZTMzZWVjOGY2YzM0MmI5ODU.

CSeq: 2 BYE

User-Agent: Sip Phone

Content-Length: 0

-------------------------------------------------

The SIPMOTOR checks if the dialog is already exist.

Mon May 28 16:42:00 2012 1153 [CCall::getDialog]

Confirmed Dialog found

Creation of a new transaction for the BYE.

Mon May 28 16:42:00 2012 21a7

[CTransaction::changeState] STATE CHANGED TO INITIAL

The BYE is a new transaction for a SIP call, in that

case, the transaction reference it is “21a7”, and the
status

is “INITIAL”.

The SIPMOTOR changes the transaction state.

Mon May 28 16:42:00 2012 21a7

[CTransaction::changeState] STATE CHANGED TO TRYING

Mon

Mon

Mon

Mon

May

May

May

May

28

28

28

28

16:42:00

16:42:00

16:42:00

16:42:00

The BYE is sent to the Call handling.

2012

 2012

2012

2012

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

Mon

May

May

May

May

May

May

May

May

May

28

28

28

28

28

28

28

28

28

16:42:00

16:42:00

16:42:00

16:42:00

16:42:00

16:42:00

16:42:00

16:42:00

16:42:00

The Call Handling answers to the SIPMOTOR.

2012

2012

2012

2012

2012

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
2012

2012

2012

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
2012

[display_ipc_out] ------------ Begin --------------Id :

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
-1

BYE

[display_ipc_out] ------------- End ----------------

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
[display_ipc_in] ------------ Begin --------------neqt :
2250 Id : -1

SUCCESSFUL

xx :

RELATIVE REQUEST : BYE

CLIR : 0

COLP : 0

CAC-CSBU info : UNKNOWN

[display_ipc_in] ------------- End ---------------

The SIPMOTOR sends the 200 ok of the BYE to the remote

SIP equipment.

Tue May 29 14:21:53 2012 SEND MESSAGE TO NETWORK

(135.118.226.39:25648 [UDP]) (BUFF LEN = 546)

----------------------utf8----------------------SIP/2.0
200 OK

Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER,

SUBSCRIBE, OPTIONS, UPDATE

Supported: replaces,timer,path,100rel

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

To: "31004" <sip:31004@oxe-

ov.alcatel.fr>;tag=ba904e80f620e0f32593273ec97e818d

From: "PC_sip_device" <sip:31024@oxe-

ov.alcatel.fr>;tag=b05ced13

Call-ID: NTEwZjI0M2VjZGY1YzExZTMzZWVjOGY2YzM0MmI5ODU.

CSeq: 2 BYE

Ed. 07

90

TG0069

Via: SIP/2.0/UDP

135.118.226.39:25648;received=135.118.226.39;branch=z9hG4bK
d87543-cf501c2f3311d050-1--d87543;rport=25648

Content-Length: 0

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

The SIPMOTOR changes the transaction state.

Mon May 28 16:42:00 2012 21a7

[CTransaction::changeState] STATE CHANGED TO COMPLETED

Mon

Mon

Mon

Mon

May

May

May

May

28

28

 28

28

16:42:00

16:42:00

16:42:00

16:42:00

The Call Handling sends a message to the SIPMOTOR to

release the “neqt” associated to

this SIP call

2012

2012

2012

2012

The SIPMOTOR acknowledge the release of the “neqt”

Mon

Mon

Mon

Mon

May

May

May

May

28

28

28

28

16:42:00

16:42:00

16:42:00

16:42:00

2012

2012

2012

2012

[display_ipc_in] ------------ Begin --------------neqt :

2250 Id : -1

SIP EQT RELEASED

[display_ipc_in] ------------- End ----------------

[display_ipc_out] ------------ Begin --------------Id :

-1

SIP_EQT_RELEASE_ACK

[display_ipc_out] ------------- End ----------------

The SIPMOTOR kills the SIP call

Mon May 28 16:42:00 2012

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
[CMotorCallManager::onIncomingEvent] killSession.

Mon May 28 16:42:00 2012 1153 [CCall::killSession]

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
The SIPMOTOR change the state of the transactions

Mon May 28 16:42:00 2012 21a5

[CTransaction::changeState] STATE CHANGED TO TERMINATED

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
...

Mon May 28 16:42:00 2012 21a7

[CTransaction::changeState] STATE CHANGED TO TERMINATED

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
11.9.2

Incoming SIP call using a SIP Trunk Group: Call Handling

point of view

Here an example of incoming call from a SIP device to an

IPtouch.

Traces option used :

>tuner km

>tuner clear-traces

>trc i

>actdbg all=off

>tuner +cpu +cpl +at hybrid=on

>actdbg sip=on abcf=on

>mtracer -a

The call arrives on the SIPMOTOR, and sending to the

Call Handling

(292779:000028)

(292779:000029)

(292779:000030)

(292779:000031)

(292779:000032)

(292779:000033)

(292779:000034)

(292779:000035)

(292779:000036)

(292779:000037)

(292779:000038)

(292779:000039)

(292779:000040)

Ed. 07

(292779:000041)

(292779:000042)

+-------------------------------------------------------
-----+

| Message received SIP ----> UA (neqt : 2250)

| INVITE : [email protected]:5060 ; user=name

| From : <PC_sip_device> [email protected]:5060 ;

user=name

| To : <"31004"> [email protected]:5060 ;
user=name

+-------------------------------------------------------
-----+

| SDP :

| @IP:port = 135.118.226.39:7888

| ALGOS :

PCMA

G729

DTMF : 101

 | DIRECTION : SEND & RECEIVE

91

| cac : false

| Prack_Required: 0

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

All the information received on the Call handling are

given by the SIPMOTOR, the SIPMOTOR has already

done an analyse and a treatment of these information.

We can see the “neqt” used to make the link between the
SIPMOTOR trace and Call Handling trace (here

2250)

The Call Handling checks the payload received.

(292779:000046) ctrl_payloads_on_reception_sdp
payloads_recu[0]=0

(292779:000047) ctrl_payloads_on_reception_sdp
payloads_recu[1]=17

(292779:000048) ctrl_payloads_on_reception_sdp
dtmf_payload 101

When a call is using a SIP Trunk Group, the call is

treated throught this SIP Trunk Group like a call a

on a T2 Trunk Group.

The Call Handling generates a SETUP message with the

information given on the INVITE, the SETUP is

different if the Trunk Group is ISDN or ABCF.

___________________________________________________________

| (292779:000128) Concatenated-Physical-Event :

| long: 177 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0

term: 0 type a5

| tei: 0 <<<< message sent : SETUP [05]

Call ref : 00 15

SENDING COMPLETE

|__________________________________________________________

| IE:[04] BEARER_CAPABILITY (l=3) 80 90 a3

| IE:[18] CHANNEL (l=2) a0 90 -> T2 : No B channel

| IE:[1c] FACILITY (l=84)

[91] Discriminator of supplementary service applications

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
[aa] NFE (l=6):

[80] Source Entity (l=1) End_PTNX

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
|

[82] Destination Entity (l=1) End_PTNX

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
[8b] Interpretation APDU (l=1): DISCARD (0)

[a1] INVOKE (l=25):

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
|

Invoke Ident. : 2ee0 (12000)

OP: ECMA RO_CALLING_NAME (0)

[80] Name presentation allowed (l=13) 'PC_sip_device'

[a1] INVOKE (l=43):

Invoke Ident. : 0001 (1)

OP: ALCATEL RO_CLASSMARKS (1)

[30] Sequence (l=30)

[80] Feature identifier (l=5) 06 04 70 1f 20

[82] Cug (l=1) 00

[ab] Sequence of Project data (l=18)

[30] Sequence (l=16)

OP :RO_CLASSMARKS_SUPPLEMENTARY_INFO_1

Ed. 07

92 (134623475)

[30] Sequence (l=10)

[80] Trunk group feature (l=5) 06 00 00 20 04

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

When the SIP message is from the SIPMOTOR to the Call

Handling, the direction is “message sent”.

On this setup all the information are present:

The calling and called number

The codecs

The RTP connection information

...

The Call Ref is identical for outgoing and incoming

messages (here Call ref : 00 15).

The “CALL PROC” is present.

___________________________________________________________

| (292779:000291) Concatenated-Physical-Event :

| long: 22 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0

term: 0 type a5

 | tei: 0 >>>> message received : CALL PROC (02) Call ref

: 00 15

|__________________________________________________________

| IE:[18] CHANNEL (l=2) a0 90 -> T2 : No B channel

|__________________________________________________________

The “ALERT” is generated for this call.

___________________________________________________________

| (292779:000294) Concatenated-Physical-Event :

| long: 101 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0

term: 0 type a5

| tei: 0 >>>> message received : ALERT (01) Call ref :

00 15

|__________________________________________________________

| IE:[1c] FACILITY (l=64)

The “CALL PROC” is present.

[91] Discriminator of supplementary service applications

[aa] NFE (l=6):

[80] Source Entity (l=1) End_PTNX

[82] Destination Entity (l=1) End_PTNX

[8b] Interpretation APDU (l=1): DISCARD (0)

[a1] INVOKE (l=28):

Invoke Ident. : 2ee1 (12001)

OP: ECMA RO_CALLED_NAME (1)

[80] Name presentation allowed (l=16) 'IPtouch 172.27.1'

[a1] INVOKE (l=20):

Invoke Ident. : 0001 (1)

ALCATEL RO_CLASSMARKS (1)

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Ed. OP:

07

93

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
|

[30] Sequence (l=7)

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
[80] Feature identifier (l=5) 06 44 7e 1f 04

TG0069

OmniPCX Enterprise

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

The ALERT has no RTP information, because the SDP on 18x

is not set to true.

The “ALERT” is transformed on a SIP message to the

SIPMOTOR, but first the Call Handling select

the good “neqt” to send the message to the SIPMOTOR.

(292779:000321)

...

(292779:000323)

(292779:000324)

(292779:000325)

(292779:000326)

(292779:000327)

(292779:000328)

SIP : [send_to_motor] ipcSend resultat : 0 sur eqt :

2250

+-------------------------------------------------------
-----+

| Message sent UA (neqt : 2250-0) ----> SIP

| Informational 180

| RELATIVE REQUEST : INVITE

| No SDP

+-------------------------------------------------------
-----+

The “CONNECT” is generated for this call.

___________________________________________________________

| (292789:000511) Concatenated-Physical-Event :

| long: 134 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0

term: 0 type a5

| tei: 0 >>>> message received : CONNECT (07) Call ref :

00 15

|__________________________________________________________

| IE:[1c] FACILITY (l=64)

[91] Discriminator of supplementary service applications

[aa] NFE (l=6):

[80] Source Entity (l=1) End_PTNX

[82] Destination Entity (l=1) End_PTNX

[8b] Interpretation APDU (l=1): DISCARD (0)

[a1] INVOKE (l=28):

 Invoke Ident. : 2ee2 (12002)

OP: ECMA RO_CONNECTED_NAME (2)

[80] Name presentation allowed (l=16) 'IPtouch 172.27.1'

[a1] INVOKE (l=20):

Invoke Ident. : 0001 (1)

OP: ALCATEL RO_CLASSMARKS (1)

[30] Sequence (l=7)

[80] Feature identifier (l=5) 06 44 7e 1f 04

| IE:[4c] CONNECTED_NUMBER (l=7) -> 00 81 Num : 31004

| [95] Locking shift. codeset : 5

| IE:[32] EI_PARTY_CATEGORY (l=1) -> EXTENSION (1)

| [9f] Non-locking shift. codeset : 7

| IE:[06] EI_IP_PAYLOADS (l=1) -> G729 Ece 1 Vad 0

| [9f] Non-locking shift. codeset : 7

| IE:[0a] EI_RTP_INFO (l=30)

-> stop_packet=0 stop_rtp=0 h323=0 wc=0 rf=0 udp=1 rqm=0

-> Transm_Bande=1 detection_Q23=1 dtmf_payload=101

-> Port RTP

= 32514, IPv4 :

172.

27. 142.

64.

07 RTCP SR = 32515, IPv4 :

-> Ed.

Port

172.

27. 142. 9464.

-> Port RTCP RR = 32515, IPv4 :

172.

27. 142.

64.

-> Port Fax

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
= 0, IPv4 :

0.

0.

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
0.

0.

TG0069

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
The “CONNECT” has RTP information, these RTP information
will be used to create the SDP.

The “CONNECT” is transformed on a SIP message to the

SIPMOTOR, but first the Call Handling

select the good “neqt” to send the message to the

SIPMOTOR.

(292789:000552)

...

(292789:000554)

(292789:000555)

(292789:000556)

(292789:000557)

(292789:000558)

(292789:000559)

(292789:000560)

(292789:000561)

(292789:000562)

(292789:000563)

(292789:000564)

(292789:000565)

(292789:000566)

(292789:000567)

SIP : [send_to_motor] ipcSend resultat : 0 sur eqt :

2250

+-------------------------------------------------------
-----+

| Message sent UA (neqt : 2250-0) ----> SIP

| Successful 200

| RELATIVE REQUEST : INVITE

+-------------------------------------------------------
-----+

| SDP :

| @IP:port = 172.27.142.64:32514

| ALGOS :

G729

DTMF : 101

| DIRECTION : SEND & RECEIVE

| AssertedAddress : <IPtouch 172.27.1> 31004@oxe-

ov.alcatel.fr:5060

| COLP

+-------------------------------------------------------
-----+

The SIPMOTOR receives the ACK from the remote SIP

equipment, and this message.

(292794:000580)

(292794:000581)

(292794:000582)

(292794:000583)

 +-------------------------------------------------------
-----+

| Message received SIP ----> UA (neqt : 2250)

| ACK

+-------------------------------------------------------
-----+

The ACK is transformed on a “CONNECT ACK”

___________________________________________________________

| (292794:000586) Concatenated-Physical-Event :

| long: 18 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0

term: 0 type a5

| tei: 0 <<<< message sent : CONNECT ACK (0f) Call ref :

00 15

|__________________________________________________________

After call establishment, the call can be released by

the OXE or by the remote SIP equipment.

Call released by the Call Handling:

The “DISCONNECT” is generated on the call.

___________________________________________________________

| (292810:000672) Concatenated-Physical-Event :

| long: 23 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0

term: 0 type a5

| tei: 0 >>>> message received : DISCONNECT [45] Call

ref : 00 15

|__________________________________________________________

| IE:[08] CAUSE (l=3) 80 90 80 -> [90] NORMAL CALL

CLEARING

|__________________________________________________________

Ed. 07

95

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

(292810:000682)

...

(292810:000684)

(292810:000685)

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
(292810:000686)

(292810:000687)

SIP : [send_to_motor] ipcSend resultat : 0 sur eqt :

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
2250

+-------------------------------------------------------
-----+

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
| Message sent UA (neqt : 2250-0) ----> SIP

| BYE

+-------------------------------------------------------
(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
-----+

(292811:000692)

(292811:000693)

(292811:000694)

(292811:000695)

(292811:000696)

(292811:000697)

The “DISCONNECT” is transformed on a SIP message to the

SIPMOTOR, but first the

Call Handling select the good “neqt” to send the message

to the SIPMOTOR.

Answer of the BYE recieved by the SIPMOTOR transmits to

the Call Handling.

+-------------------------------------------------------
-----+

| Message received SIP ----> UA (neqt : 2250)

| Successful 200

| RELATIVE REQUEST : BYE

| No SDP

+-------------------------------------------------------
-----+

Answer of the BYE is transformed to a Call Handling

message by a “RELEASE”.

___________________________________________________________

| (292811:000699) Concatenated-Physical-Event :

| long: 23 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0

term: 0 type a5

| tei: 0 <<<< message sent : RELEASE [4d]

Call ref : 00 15

|__________________________________________________________

| IE:[08] CAUSE (l=3) 80 90 80 -> [90] NORMAL CALL

CLEARING

|__________________________________________________________

Acknowledge of the “RELEASE” by a “REL COMP”.

___________________________________________________________

| (292811:000705) Concatenated-Physical-Event :

| long: 23 desti: 0 source: 0 cryst: 19 cpl: 0 us: 0

term: 0 type a5

| tei: 0 >>>> message received : REL COMP [5a] Call ref

: 00 15

|__________________________________________________________

| IE:[08] CAUSE (l=3) 80 90 80 -> [90] NORMAL CALL

 CLEARING

|__________________________________________________________

After the “REL COMP”, the call is completely ended on

Call Handling side.

According to the problem, more options can be used on

the Call Handling trace, due to them, more

information are displayed. Here it is an example with

the minimum of options to see the exchanges between

the SIPMOTOR and the Call Handling.

It is important to understand the link between SIPMOTOR

trace and Call Handling trace to make a minimum

of analyses before to open a Service Request.

Ed. 07

96

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.9.3

Incoming SIP call in case of SIP extension: SIPMOTOR

point of view

Here an example of incoming call from a SIP extension to

an IPtouch.

Tue Jun 26 08:03:05 2012 RECEIVE MESSAGE FROM NETWORK

(135.118.226.21:61618 [UDP])

----------------------utf8----------------------INVITE
sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 135.118.226.21:61618;branch=z9hG4bK-

d87543-9c72747c0d38bb69-1--d87543-;rport

Max-Forwards: 70

Contact: <sip:[email protected]:61618>

To: "31004"<sip:[email protected]>

From: "PC_sip_extenstion"<sip:31023@oxe-
ov.alcatel.fr>;tag=c850be7c

Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.

CSeq: 1 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,

MESSAGE, SUBSCRIBE, INFO

Content-Type: application/sdp

User-Agent: SIP Phone

Content-Length: 317

v=0

o=- 5 2 IN IP4 135.118.226.21

s=SIP Phone

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
c=IN IP4 135.118.226.21

t=0 0

m=audio 46194 RTP/AVP 8 18 101

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
a=fmtp:18 annexb=no

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
a=sendrecv

------------------------------------------------Ed. 07

97

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The information “RECEIVE MESSAGE FROM NETWORK

(135.118.226.21:61618[UDP])” is important to

know that the call is an incoming one from the SIP

equipment 135.118.226.21 in UDP.

The OXE checks the Call-Id to know if this INVITE is an

INVITE or a REINVITE.

Tue Jun 26 08:03:05 2012 11ef [CCall::getDialog]

Confirmed Dialog is not found (ID = ;c850be7c)

Tue Jun 26 08:03:05 2012 11ef [CCall::getDialog] Initial

Dialog Server not found

Here, it is an INVITE, because the dialog is not found.

The transaction and the dialog are put in place.

Tue Jun 26 08:03:05 2012 210c

[CTransaction::changeState] STATE CHANGED TO INITIAL

...

Tue Jun 26 08:03:05 2012 15fd

[CDialog::onTransactionState(pTrans = 210c,
previousState = Terminated, currentState

= Initial, reason = None]

Here, the transaction reference is “210c” and the dialog

reference is “15fd”.

The transaction status is changed, because the dialog is

initiated.

Tue Jun 26 08:03:05 2012 210c

[CTransaction::changeState] STATE CHANGED TO PROCEEDING

Tue Jun 26 08:03:05 2012 210c

[CTransaction::changeState] notifying the parent dialog

When a transaction is linked to a dialog, the

transaction changed from INITIAL to PROCEEDING.

The SIPMOTOR generates the 100 Trying.

Tue Jun 26 08:03:05 2012 SEND MESSAGE TO NETWORK

(135.118.226.21:61618 [UDP]) (BUFF LEN = 350)

----------------------utf8----------------------SIP/2.0
100 Trying

To: "31004" <sip:[email protected]>

From: "PC_sip_extenstion" <sip:31023@oxe-

ov.alcatel.fr>;tag=c850be7c

Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.

CSeq: 1 INVITE

Via: SIP/2.0/UDP
135.118.226.21:61618;received=135.118.226.21;branch=z9hG4bK
d87543-9c72747c0d38bb69-1--d87543;rport=61618

Content-Length: 0

-------------------------------------------------

The 100 Trying is generated by the SIPMOTOR.

The SIPMOTOR checks the Session Timer for the call.

 Tue Jun 26 08:03:05

(Server, UA)

Tue Jun 26 08:03:05

reception)

Tue Jun 26 08:03:05

Tue Jun 26 08:03:05

Ed. 07

2012 [CSessionTimerContext::CSessionTimerContext] New

CSessionTimerContext from request

2012 [CSessionTimerContext::updateAfterRefreshReception]
Update CSessionTimerContext (refresh

2012 [CSessionTimerContext::updateSessionExpires]
Session-Expires updated : 0

2012 [CSessionTimerContext::setRefreshMethod] Allow

refreshMethod=INVITE

98

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

In this case, the SIP equipment doesn‟t send “Session

timer” information because the value is 0 (updated :

0).

The SIPMOTOR makes the link between the transaction, the

branch and the Cseq number.

Tue Jun 26 08:03:05 2012 15fd [CDialog::addTransaction]

added transaction 210c with branch z9hG4bK-
d875439c72747c0d38bb69-1--d87543-, with CSeq 1

The “branch” is a parameter added to the “via” to

identify it. Regarding rfc3261, all the branch

values must start with “z9hG4bK”.

The CSeq is used to indentify and to order a

transaction, it consists of a sequence number and a
method.

The SIPMOTOR checks for which OXE equipment the call is

from.

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Tue

Tue

Tue

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Jun

Jun

Jun

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Jun

Jun

Jun

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Jun

Jun

Jun

Jun

Jun

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26

26

26

26

26

26

26

26

26

26

26

26

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

2012

2012

2012

2012

2012

2012

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2012

2012

2012

2012

2012

[isDomainFromGwExt] Host from request is :

172.27.141.151.

[isDomainFromGwExt] User from request is : 31023

[domain not from an External Gateway.

11ef[CMotorCall::setFilterUsedMode] To be traced = 0

11ef[CMotorCall::initOfUserType] values are reseted

[getFromHeader] displayName="PC_sip_extenstion".

[getFromHeader] [email protected].

 [getFromHeader] clirPresent=0.

[isAddrInDico] user=31023 host=oxe-ov.alcatel.fr

[isUserInDico] [email protected]

[isUserInDico] found in the dictionnary.

[isAddrInDico] seplos station OK

Here, we can see that the call is from a SEPLOS station.

The SIPMOTOR checks the number of licenses available.

Tue Jun 26 08:03:05 2012

11ef[CMotorCall::methodInviteReceived] nb available
licenses=25

Here the number of licenses is 25, that means, 25 calls

are possible on SIP using a SIP Trunk Group or

SEPLOS users

The SIPMOTOR checks if the IP address recieved is

managed on an IP domain.

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

...

Tue

Jun

Jun

Jun

Jun

Jun

Jun

Jun

Jun

26

26

26

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26

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26

08:03:05

08:03:05

08:03:05

08:03:05

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
08:03:05

08:03:05

08:03:05

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
08:03:05

2012

2012

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
2012

2012

2012

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
2012

2012

2012

Jun 26 08:03:05 2012

The recevied host 135.118.226.21

Trying to find the ip address in domain list

The entry dom : 142 add_type=1

The entry dom ip low :172.27.141.165

The entry ipaddress from low :135.118.226.21

The entry compare :1

The entry compare 2 :0

iplink_is_good_range_for_reg

The user domain is

142

Here, the IP address of the SIP equipment corresponds to

the IP domain 142.

If the IP address doesn‟t match an IP domain, the

SIPMOTOR returns:

Tue Jun 26 08:03:05 2012

The user is ipadd

not in any Domain range return state as -1

The SIPMOTOR checks the SDP received on the INVITE.

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Jun 26 08:03:05

Jun 26 08:03:05

Jun 26 08:03:05

Jun 26 08:03:05

Jun 26 08:03:05

Ed. 07

Jun 26 08:03:05

Jun 26 08:03:05

Jun 26 08:03:05

2012

2012

2012

2012

2012

2012

2012

2012

[checkSdpValidity] Media 0 type 1 contains 3 formats.

[checkSdpValidity] Format : 8.

11ef[CMotorCall::isCryptoAuthorized] user crypto=0.

 [convertSdpIntoTsdp] No Direction in the session part.

[convertSdpIntoTsdp] Check the direction in Session part

- result:0.

99

[convertSdpIntoTsdp] media AUDIO detected (previous

crypto=0).

[convertAudioMedia] The audio media contains 3

format(s).

[convertAudioMedia] Format 0 is 8.

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The SDP contains in this SDP three formats of medias (8,

18 and 101), the direction is “sendrecv” meaning

in both direction and the IP address of connection is

135.118.226.21.

The message to Call Handling is prepared and sent to it.

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Jun

Jun

Jun

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Jun

Jun

Jun

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Jun

Jun

Jun

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Jun

Jun

Jun

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Jun

Jun

Jun

Jun

Jun

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Jun

Jun

Jun

Jun

26

26

26

26

26

26

26

26

26

26

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26

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

 08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

Ed. 07

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

2012

11ef[CMotorCall::sendLgEvtSipCreate] Event sent on eqt :

2066

** SEPLOS **

[display_ipc_out] ------------ Begin --------------Id :

-1

INVITE

REQUEST URI : <> [email protected]:5060 ;

user=name

FROM : <PC_sip_extenstion> [email protected]:5060

; user=name

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
TO : <"31004"> [email protected]:5060 ; user=name

CAC : 0

CAC ADDRESS :

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
CAC-CSBU info : UNKNOWN

CLIR : 0

Prack Required : 0

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Allow Update : 0

SDP :

ADDRESS : 135.118.226.21 :46194

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
ALGOS :

PCMA

G729

101

DIRECTION : SEND & RECEIVE

crypto index : 0

N_GW_EXT : -1

[display_ipc_out] ------------- End ----------------

100

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

The call is sent to the Call handling on neqt 2066,

regarding the type of SIP equipment detected by the

SIPMOTOR, some information are added or not on this

message.

All the information about this call are sent to the

Stand-By CPU.

Tue Jun 26 08:03:05 2012 SendToSipgwCpuSec: Message sent

to the STAND-BY CPU

Tue Jun 26 08:03:05 2012 [receiveInviteMessage] send

RemoteSdp to the StandBy.

Tue Jun 26 08:03:05 2012 SendToSipgwCpuSec: Message sent

to the STAND-BY CPU

The information are sent to the Stand-By, like this, in

case of bascul the SIP call will not be lost and known

on the new main CPU

The Call handling send back an answer for this INVITE.

Tue

Tue

Tue

Tue

Tue

Tue

Jun

Jun

Jun

Jun

Jun

Jun

26

26

26

26

26

26

08:03:05

08:03:05

08:03:05

 08:03:05

08:03:05

08:03:05

2012

2012

2012

2012

2012

2012

[display_ipc_in] ------------ Begin --------------neqt :

2066 Id : 1

INFORMATIONAL

xx :

RELATIVE REQUEST : INVITE

[display_ipc_in] ------------- End ----------------

A “100 Trying” is sent by the Call Handling , but

ignored by the SIPMOTOR.

Tue Jun 26 08:03:05 2012 [onIncomingEvent] INFORMATIONAL

arrived.

Tue Jun 26 08:03:05 2012 [onIncomingEvent] 100 TRYING

ignored.

This 100 Trying generated by the Call Handling is used

by it to assign a “session” number for this call on the

Call Handling side, but not used by the SIPMOTOR

The Call handling sends back an answer for this INVITE.

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Jun

Jun

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Jun

Jun

Jun

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Jun

Jun

Jun

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Jun

Jun

Jun

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Jun

26

26

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
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08:03:05

08:03:05

08:03:05

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08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

08:03:05

2012

2012

2012

2012

2012

2012

2012

2012

2012

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2012

2012

2012

[display_ipc_in] ------------ Begin --------------neqt :

2066 Id : 1

INFORMATIONAL

xx :

80

RELATIVE REQUEST : INVITE

SDP :

ADDRESS : 172.27.143.131 :32584

ALGOS :

G729

101

DIRECTION : SEND & RECEIVE

crypto index : 0

 [display_ipc_in] ------------- End ----------------

A “180 Ringing” is sent by the Call Handling with SDP,

for the moment, on a 18X message, the Call Handling

will put everytime a SDP, no possibility to disable it.

The SIPMOTOR checks if the SDP given is compatible with

the SDP received in the INVITE.

1340690585 -> Tue Jun 26

Tue Jun 26 08:03:05 2012

Tue Jun 26 08:03:05 2012

Tue Jun 26 08:03:05 2012

Tue Jun 26 08:03:05 2012

Tue Jun 26 08:03:05 2012

Tue Jun 26 08:03:05 2012

Tue Jun 26 08:03:05 2012

Tue Jun 26 08:03:05 2012

Ed. 07

a=sendrecv

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

08:03:05 2012 11ef[CMotorCall::makeResponseSdp] Audio

media.

11ef[CMotorCall::appendAudioAttributToMedia] Direction:
0.

11ef[CMotorCall::appendAudioAttributToMedia] format 101

11ef[CMotorCall::makeResponseSdp]
fromSdp.getMediaDesciprionCount :1

[sameCodec] accepted Format : 18.

[sameCodec] requested Format : 8.

[sameCodec] requested Format : 18.

[sameCodec] same Format.

11ef[CMotorCall::mediaAccepted] Media accepted: m=audio

32584 RTP/AVP 18 101

101

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The codecs from the INVITE were 8 and 18, on the answer
we have 18, in that case the call is accepted by

SIPMOTOR for SDP point of view.

The Call handling sends back an answer for this INVITE.

1340690585 -> Tue Jun 26 08:03:05 2012 SEND MESSAGE TO

NETWORK (135.118.226.21:61618 [UDP]) (BUFF LEN = 827)

----------------------utf8----------------------SIP/2.0
180 Ringing

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER,
SUBSCRIBE, OPTIONS, UPDATE

Contact: sip:oxe-ov.alcatel.fr

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
User-Agent: OmniPCX Enterprise R10.0 j1.410.45

Content-Type: application/sdp

To: "31004" <sip:31004@oxe-

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
ov.alcatel.fr>;tag=05b5888d18d4e78f3554a55dadeefb08

From: "PC_sip_extenstion" <sip:31023@oxe-

ov.alcatel.fr>;tag=c850be7c

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.

CSeq: 1 INVITE

Via: SIP/2.0/UDP
135.118.226.21:61618;received=135.118.226.21;branch=z9hG4bK
d87543-9c72747c0d38bb69-1--d87543;rport=61618

Content-Length: 243

v=0

o=OXE 1340690585 1340690585 IN IP4 172.27.141.151

s=abs

c=IN IP4 172.27.143.131

t=0 0

m=audio 32584 RTP/AVP 18 101

a=sendrecv

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=ptime:20

a=maxptime:40

a=rtpmap:101 telephone-event/8000

-------------------------------------------------

For each SIP call event, a message is send to the Stand-

By CPU.

Tue Jun 26 08:03:05 2012 [receiveInformationalEvent]

UpdateContext send on the StandBy.

The Call handling sends a new answer for this INVITE.

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Jun

Jun

Jun

Jun

Jun

Jun

Jun

Jun

Jun

Jun

 Jun

Jun

Jun

Jun

Jun

Jun

26

26

26

26

26

26

26

26

26

26

26

26

26

26

26

26

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

08:03:08

2012

2012

2012

2012

2012

2012

2012

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
2012

2012

2012

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
2012

2012

2012

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
2012

2012

2012

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
[display_ipc_in] ------------ Begin --------------neqt :
2066 Id : 1

SUCCESSFUL

xx :

RELATIVE REQUEST : INVITE

CLIR : 0

COLP : 1

CAC-CSBU info : UNKNOWN

SDP :

ADDRESS : 172.27.142.64 :32514

ALGOS :

G729

101

DIRECTION : SEND & RECEIVE

crypto index : 0

[display_ipc_in] ------------- End ----------------

A “200 ok” is sent to the SIPMOTOR with SDP

Ed. 07

102

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The SIPMOTOR checks if the SDP given is compatible with

the SDP received in the INVITE.

1340690588 -> Tue Jun 26 08:03:08 2012

11ef[CMotorCall::makeResponseSdp] Audio media.

Tue Jun 26 08:03:08 2012

11ef[CMotorCall::appendAudioAttributToMedia] Direction:
0.

Tue Jun 26 08:03:08 2012

11ef[CMotorCall::appendAudioAttributToMedia] format 101

Tue Jun 26 08:03:08 2012

11ef[CMotorCall::makeResponseSdp]
fromSdp.getMediaDesciprionCount :1

Tue Jun 26 08:03:08 2012 [sameCodec] accepted Format :

18.

Tue Jun 26 08:03:08 2012 [sameCodec] requested Format :

8.

Tue Jun 26 08:03:08 2012 [sameCodec] requested Format :

18.

Tue Jun 26 08:03:08 2012 [sameCodec] same Format.

Tue Jun 26 08:03:08 2012 11ef[CMotorCall::mediaAccepted]

Media accepted: m=audio 32514 RTP/AVP 18 101

a=sendrecv

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=ptime:20

a=maxptime:40

 a=rtpmap:101 telephone-event/8000

The codecs from the INVITE were 8 and 18, on the answer
we have 18, in that case the call is accepted by

SIPMOTOR for SDP point of view.

The SIPMOTOR is changing the status of the dialog.

Tue Jun 26 08:03:08 2012 15fd [CDialog::createResponse]

create a CONFIRMED dialog

Due to this, the dialog reference and transaction

reference are changed (internal SIPMOTOR functionning).

Tue Jun 26 08:03:08 2012 15fe [CDialog::CDialog] look

for the transaction #0, transaction key = z9hG4bK-
d875439c72747c0d38bb69-1--d87543Tue Jun 26 08:03:08 2012
15fe [CDialog::CDialog] copy the transaction #0,
transaction key = z9hG4bK-d875439c72747c0d38bb69-1--
d87543Tue Jun 26 08:03:08 2012 210d
[CTransaction::CTransaction] Transaction is cloned in 4
state

The dialog reference is changed form “15fd” to “15fe”.

The transaction reference is changed from “210c” to

“210d”.

The SIPMOTOR is changing the stus of the dialog.

Tue Jun 26 08:03:08 2012 SEND MESSAGE TO NETWORK

(135.118.226.21:61618 [UDP]) (BUFF LEN = 984)

----------------------utf8----------------------SIP/2.0
200 OK

Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER,

SUBSCRIBE, OPTIONS, UPDATE

Contact: sip:oxe-ov.alcatel.fr

Supported: replaces,timer,path,100rel

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

Session-Expires: 1800;refresher=uas

P-Asserted-Identity: "IPtouch 172.27.142.64"

<sip:[email protected];user=phone>

Content-Type: application/sdp

To: "31004" <sip:31004@oxe-

ov.alcatel.fr>;tag=05b5888d18d4e78f3554a55dadeefb08

From: "PC_sip_extenstion" <sip:31023@oxe-

ov.alcatel.fr>;tag=c850be7c

Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.

CSeq: 1 INVITE

Via: SIP/2.0/UDP
135.118.226.21:61618;received=135.118.226.21;branch=z9hG4bK
d87543-9c72747c0d38bb69-1--d87543;rport=61618

Content-Length: 242

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
v=0

o=OXE 1340690585 1340690586 IN IP4 172.27.141.151

s=abs

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
c=IN IP4 172.27.142.64

t=0 0

m=audio 32514 RTP/AVP 18 101

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
a=sendrecv

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
a=ptime:20

a=maxptime:40

a=rtpmap:101

Ed. 07 telephone-event/8000

-------------------------------------------------

103

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

The 200ok sent to the remote SIP equipment is generated

with information from the INVITE received and

from the 200ok answer from the Call Handling.

The SIPMOTOR is changing the status of the transaction.

Tue Jun 26 08:03:08 2012 210d

[CTransProceedingState::createResponse] Final :
Transaction changes to Completed

state

The retransmission timers are started.

Tue Jun 26 08:03:08 2012 210d [CTransaction::startTimer]

Timer G is started (delay = 500 ms)

Tue Jun 26 08:03:08 2012 210d [CTransaction::startTimer]

Timer H is started (delay = 32000 ms)

The SIPMOTOR receives a ACK for the 200ok.

Tue Jun 26 08:03:08 2012 RECEIVE MESSAGE FROM NETWORK

(135.118.226.21:61618 [UDP])

----------------------utf8----------------------ACK
sip:oxe-ov.alcatel.fr SIP/2.0

Via: SIP/2.0/UDP 135.118.226.21:61618;branch=z9hG4bK-

d87543-cc14ac1776189458-1--d87543-;rport

Max-Forwards: 70

Contact: <sip:[email protected]:61618>

To: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=05b5888d18d4e78f3554a55dadeefb08

From: "PC_sip_extenstion"<sip:31023@oxe-
ov.alcatel.fr>;tag=c850be7c

Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.

CSeq: 1 ACK

User-Agent: SIP Phone

Content-Length: 0

-------------------------------------------------

The SIPMOTOR is changing the status of the transaction.

Tue Jun 26 08:03:08 2012 210d

[CTransaction::changeState] STATE CHANGED TO TERMINATED

The retransmission timers are freed.

Tue Jun 26 08:03:08 2012 210d

[CTransaction::freeTimerToken] Timer G is freed

Tue Jun 26 08:03:08 2012 210d

[CTransaction::freeTimerToken] Timer H is freed

The SIPMOTOR is changing the status of the dialog.

 Tue Jun 26 08:03:08 2012 15fe

[CDialog::receiveAckRequest] the INVITE request is
terminated

The ACK is sent to the Call Handling.

Tue

Tue

Tue

Tue

Jun

Jun

Jun

Jun

26

26

26

26

08:03:08

08:03:08

08:03:08

08:03:08

2012

2012

2012

2012

[display_ipc_out] ------------ Begin --------------Id :

1

ACK

[display_ipc_out] ------------- End ----------------

After call establishment, the call can be released by

the OXE or by the remote SIP equipment.

Call released by the OXE:

Tue

Tue

Tue

Tue

Jun

Jun

Jun

Jun

26 08:03:10

26 08:03:10

26 08:03:10

26 08:03:10

Ed. 07

The BYE is sent from the Call Handling.

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
2012

2012

2012

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
2012

[display_ipc_in] ------------ Begin --------------neqt :

2066 Id : 1

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
BYE

[display_ipc_in] ------------- End ---------------104

TG0069

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

Creation of a new transaction for the BYE.

Tue Jun 26 08:03:10 2012 2110

[CTransaction::changeState] STATE CHANGED TO INITIAL

The BYE is a new transaction for a SIP call, in that

case, the transaction reference it is “2110”, and the
status

is “INITIAL”.

The BYE is sent to the remote SIP equipment.

Tue Jun 26 08:03:10 2012 SEND MESSAGE TO NETWORK

(135.118.226.21:61618 [UDP]) (BUFF LEN = 454)

----------------------utf8----------------------BYE
sip:[email protected]:61618 SIP/2.0

Supported: replaces,timer,path,100rel

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

To: sip:[email protected];tag=c850be7c

From: "31004" <sip:31004@oxe-

ov.alcatel.fr>;tag=05b5888d18d4e78f3554a55dadeefb08

Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.

CSeq: 716266225 BYE

Via: SIP/2.0/UDP
172.27.141.151;branch=z9hG4bK2385fb34fcefc38c24fa6848df37e9

Max-Forwards: 70

Content-Length: 0

-------------------------------------------------

The SIPMOTOR changes the transaction state.

Tue Jun 26 08:03:10 2012 2110

[CTransaction::changeState] STATE CHANGED TO TRYING

The retransmission timers are started.

Tue Jun 26 08:03:10 2012 2110 [CTransaction::startTimer]

Timer E is started (delay = 500 ms)

Tue Jun 26 08:03:10 2012 2110 [CTransaction::startTimer]

Timer F is started (delay = 16000 ms)

The 200ok of the BYE request is received from the remote

SIP equipment.

Tue Jun 26 08:03:10 2012 RECEIVE MESSAGE FROM NETWORK

(135.118.226.21:61618 [UDP])

----------------------utf8----------------------SIP/2.0
200 OK

Via: SIP/2.0/UDP
172.27.141.151;branch=z9hG4bK2385fb34fcefc38c24fa6848df37e9

 Contact: <sip:[email protected]:61618>

To: <sip:[email protected]>;tag=c850be7c

From: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=05b5888d18d4e78f3554a55dadeefb08

Call-ID: MzBlMzgzNjY5NDg2NmE0NTRiMGYyYjMyOThjZmY4MWU.

CSeq: 716266225 BYE

User-Agent: SIP Phone

Content-Length: 0

-------------------------------------------------

The SIPMOTOR changes this transaction state.

Tue Jun 26 08:03:10 2012 2110

[CTransaction::changeState] STATE CHANGED TO COMPLETED

The retransmission timers are freed.

Tue Jun 26 08:03:10 2012 2110

[CTransaction::freeTimerToken] Timer E is freed

Tue Jun 26 08:03:10 2012 2110

[CTransaction::freeTimerToken] Timer F is freed

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Jun 26 08:03:10

Jun 26 08:03:10

Jun 26 08:03:10

Jun 26 08:03:10

Jun Ed.

26 07

08:03:10

Jun 26 08:03:10

Jun 26 08:03:10

The 200ok of the BYE request is sent to the Call

Handling.

2012

2012

2012

2012

2012

2012

2012

** SEPLOS **

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
[sendLgEvtSip] Event sent on eqt : 2066 Id :1

[display_ipc_out] ------------ Begin --------------Id :

1

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
SUCCESSFUL

105

xx :

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
0

RELATIVE REQUEST : BYE

TG0069

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Tue

Tue

Tue

Tue

Jun

Jun

Jun

Jun

26

26

26

26

08:03:10

08:03:10

08:03:10

08:03:10

The Call Handling sent a message to the SIPMOTOR to

release the “neqt” associated to

this SIP call

2012

2012

2012

2012

The SIPMOTOR acknowledge the release of the “neqt”

Tue

Tue

Tue

Tue

Jun

Jun

Jun

Jun

26

26

26

26

08:03:10

08:03:10

08:03:10

08:03:10

2012

2012

2012

2012

[display_ipc_in] ------------ Begin --------------neqt :

2066 Id : 1

 SIP EQT RELEASED

[display_ipc_in] ------------- End ----------------

[CMotorCallManager::onIncomingEvent] The call with eqt:

2066 has released its equipment.

[CMotorCallManager::onIncomingEvent] state =
TERMINATED_STATE.

11ef[CMotorCall::unRegister] Remove eqt : 2066 diag : 1

from the map.

[CMotorCallManager::eraseCallwithEqt] erase 2066 1.

The SIPMOTOR kills the SIP call

Tue Jun 26 08:03:10 2012

[CMotorCallManager::onIncomingEvent] killSession.

Tue Jun 26 08:03:10 2012 11ef [CCall::killSession]

The SIPMOTOR changes the state of the transactions

Tue Jun 26 08:03:10 2012 210c

[CTransaction::changeState] STATE CHANGED TO TERMINATED

...

Tue Jun 26 08:03:10 2012 2110

[CTransaction::changeState] STATE CHANGED TO TERMINATED

Call released by the remote SIP equipment:

The BYE is received from the remote SIP equipment.

Tue Jun 26 08:03:10 2012 RECEIVE MESSAGE FROM NETWORK

(135.118.226.21:61618 [UDP])

----------------------utf8----------------------BYE
sip:oxe-ov.alcatel.fr SIP/2.0

Via: SIP/2.0/UDP 135.118.226.21:61618;branch=z9hG4bK-

d87543-c47926131a084707-1--d87543-;rport

Max-Forwards: 70

Contact: <sip:[email protected]:61618>

To: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=efa4b05316a486724541975cb22707d1

From: "PC_sip_extenstion"<sip:31023@oxe-
ov.alcatel.fr>;tag=c55fb830

Call-ID: MzIwMmRjNGI3YTE3ZjkwZTE0ODE4Y2IzZGU1ZTdjZDM.

CSeq: 2 BYE

User-Agent: SIP Phone

Content-Length: 0

-------------------------------------------------

The SIPMOTOR checks if the dialog is already exist.

Tue Jun 26 08:03:10 2012 11ef [CCall::getDialog]

Confirmed Dialog found

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Creation of a new transaction for the BYE.

Tue Jun 26 08:03:10 2012 2110

[CTransaction::changeState] STATE CHANGED TO INITIAL

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Ed. 07

106

TG0069

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
The BYE is a new transaction for a SIP call, in that
case, the transaction reference it is “21a7”, and the
status

is “INITIAL”.

The SIPMOTOR changes the transaction state.

Tue Jun 26 08:03:10 2012 2110

[CTransaction::changeState] STATE CHANGED TO TRYING

Tue

Tue

Tue

Tue

Jun

Jun

Jun

Jun

26

26

26

26

08:03:10

08:03:10

08:03:10

08:03:10

The BYE is sent to the Call handling.

2012

2012

2012

2012

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Tue

Jun

Jun

Jun

Jun

Jun

Jun

Jun

Jun

Jun

26

26

26

26

 26

26

26

26

26

08:03:10

08:03:10

08:03:10

08:03:10

08:03:10

08:03:10

08:03:10

08:03:10

08:03:10

[display_ipc_out] ------------ Begin --------------Id :

-1

BYE

[display_ipc_out] ------------- End ----------------

The Call Handling answers to the SIPMOTOR.

2012

2012

2012

2012

2012

2012

2012

2012

2012

[display_ipc_in] ------------ Begin --------------neqt :

2266 Id : -1

SUCCESSFUL

xx :

RELATIVE REQUEST : BYE

CLIR : 0

COLP : 0

CAC-CSBU info : UNKNOWN

[display_ipc_in] ------------- End ---------------

The SIPMOTOR sends the 200 ok of the BYE to the remote

SIP equipment.

Tue Jun 26 08:03:10 2012 SEND MESSAGE TO NETWORK

(135.118.226.39:25648 [UDP]) (BUFF LEN = 546)

----------------------utf8----------------------SIP/2.0
200 OK

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER,
SUBSCRIBE, OPTIONS, UPDATE

Supported: replaces,timer,path,100rel

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
User-Agent: OmniPCX Enterprise R10.0 j1.410.45

To: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=efa4b05316a486724541975cb22707d1

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
From: "PC_sip_extenstion"<sip:31023@oxe-
ov.alcatel.fr>;tag=c55fb830

Call-ID: MzIwMmRjNGI3YTE3ZjkwZTE0ODE4Y2IzZGU1ZTdjZDM.

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
CSeq: 2 BYE

Via: SIP/2.0/UDP
135.118.226.39:25648;received=135.118.226.39;branch=z9hG4bK
d87543-cf501c2f3311d050-1--d87543;rport=25648

Content-Length: 0

-------------------------------------------------

The SIPMOTOR changes the transaction state.

Tue Jun 26 08:03:10 2012 2110

[CTransaction::changeState] STATE CHANGED TO COMPLETED

Tue

Tue

Tue

Tue

Jun

Jun

Jun

Jun

26

26

26

26

08:03:10

08:03:10

08:03:10

08:03:10

Ed. 07

The Call Handling sends a message to the SIPMOTOR to

release the “neqt” associated to

this SIP call

2012

2012

2012

2012

[display_ipc_in] ------------ Begin --------------neqt :

2266 Id : -1

SIP EQT RELEASED

[display_ipc_in] ------------- End ---------------107

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

The SIPMOTOR acknowledge the release of the “neqt”

Tue

Tue

Tue

Tue

Jun

Jun

Jun

Jun

 26

26

26

26

08:03:10

08:03:48

08:03:48

08:03:48

2012

2012

2012

2012

Session Iniation Protcol (SIP)

[CMotorCallManager::onIncomingEvent] The call with eqt:

2066 has released its equipment.

[CMotorCallManager::onIncomingEvent] state =
TERMINATED_STATE.

11fc[CMotorCall::unRegister] Remove eqt : 2066 diag : 1

from the map.

[CMotorCallManager::eraseCallwithEqt] erase 2066 1

The SIPMOTOR kills the SIP call

Tue Jun 26 08:03:10 2012

[CMotorCallManager::onIncomingEvent] killSession.

Tue Jun 26 08:03:10 2012 11ef [CCall::killSession]

The SIPMOTOR change the state of the transactions

Tue Jun 26 08:03:10 2012 210c

[CTransaction::changeState] STATE CHANGED TO TERMINATED

...

Tue Jun 26 08:03:10 2012 2110

[CTransaction::changeState] STATE CHANGED TO TERMINATED

11.9.4

Incoming SIP call in case of SIP extension: Call

Handling point of view

Here an example of incoming call from a SIP extension to

an IPtouch.

Traces option used :

>tuner km

>tuner clear-traces

>trc i

>actdbg all=off

>tuner +cpu +cpl +at hybrid=on

>actdbg sip=on csip=on

>mtracer -a

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
The call arrives on the SIPMOTOR, and sending to the
Call Handling

(600095:000062)

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
(600095:000063)

(600095:000064)

(600095:000065)

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
(600095:000066)

(600095:000067)

(600095:000068)

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
(600096:000069)

(600096:000070)

(600096:000071)

(600096:000072)

(600096:000073)

(600096:000074)

(600096:000075)

(600096:000076)

Ed. 07

(600096:000077)

(600096:000078)

CSIP @@@@@@@@@@@@@@@@@@@@@@@@@@@@@ 02066 activated

@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@

CSIP_receiveSipMsg

+-------------------------------------------------------
-----+

| Message received SIP ----> UA (neqt : 2066)

| INVITE : [email protected]:5060 ; user=name

| From : <PC_sip_extenstion> 31023@oxe-

ov.alcatel.fr:5060 ; user=name

| To : <"31004"> [email protected]:5060 ;
user=name

+-------------------------------------------------------
-----+

| SDP :

| @IP:port = 135.118.226.21:46194

| ALGOS :

PCMA

G729

DTMF : 101

| DIRECTION : SEND & RECEIVE

108

| cac : false

| Prack_Required: 0

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

In case of SIP Extension, all The call Handling

treatment for the call starts by the message “CSIP”, for
SIP

extension point of view.

On the first line, the information “02066 activated” is

used to inform that the Call Handling starts the

treatment of the SIP extension with the neqt 2066.

The Call Handling checks if a session is already opened

for this SIP extension user.

 (600096:000087)

(600096:000088)

(600096:000089)

(600096:000090)

(600096:000091)

..CSIPMsgSipInvite::getSession

....CSIP_getSessionFromRequestURI

......Didn't retrieve session for requestUri 31004

....CSIP_getFreeSession

......Got free session 1 for ChId 80

CSIP_INVITE_WAIT_STATUS_CH_ID

In that case, no session opened, the Call Handling

assigns to this call the session number 1, for a second

call (if the first call is still up) the session will be
2, etc...

The Call Handling generates a 100 Trying for this

session

(600096:000094)

(600096:000095)

(600096:000096)

(600096:000097)

(600096:000098)

(600096:000099)

(600096:000100)

(600096:000101)

(600096:000102)

......CSIPSession#1ChId#80::sendSipInformational

........CSIPSession#1ChId#80::emitMsgToSIPMotor

..........SIP_INFORMATIONAL sent

+-------------------------------------------------------
-----+

| Message sent UA (neqt : 2066-1) ----> SIP

| Informational 100

| RELATIVE REQUEST : INVITE

| No SDP

+-------------------------------------------------------
-----+

This 100 Trying will not take into account by the

SIPMOTOR, it is used only to start the session on the
Call

handling side.

Getting the SDP information received

(600096:000121)

(600096:000122)

(600096:000123)

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
(600096:000124)

(600096:000125)

(600096:000126)

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
CSIP_tradKey chId=128 CSIP_START_CALL

CSIP_analyzeSdp 135.118.226.21:46194 DTMF=101

SIP_SENDRECV

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G_711_A/G_729_A -> G_711_A/G_729_A

CSIP_tradKey -> cnx_create_tab(0, -1,

135.118.226.21:46194)

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
CSIP_tradKey kindofkey=VSYST (6) cokey=17

CSIP_sendInfoCs : No call server informations

authorization.

This 100 Trying will not take into account by the

SIPMOTOR, it is used only to start the session on the
Call

handling side.

Ed. 07

109

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Analyse of the SDP information

(600096:000136) put_rtp_info end 2066 local.wc=0

distant.wc=0

(600096:000137) sip_ems_with_rfc2833--
>disa_for_remote_ext=0

(600096:000138) sip_ems_with_rfc2833-->Result=0

(600096:000139) Exist_RCL_link--
>Result=0,dtmf_direction=1

(600096:000141) SIP: mise a jour VPN

(600096:000142) dtmf_to_vpn_from_abc :
dtmf_payload(2066)=101

(600096:000143) dtmf_to_vpn_from_abc : !LIEN_VPN

(600096:000144) Marhaban bikom dans le monde SIP :

dtmf_payload(2066)= 101

(600096:000145) CSIP_isNwkCallWithSeplos neqt 2066 abc

-1 vpn -1 result 0

(600096:000146) is_ems_ext_gw-->neqt=2066,Result=0

(600096:000147) send_cpl_connect_rtp_direct--
>dtmf_direction=1

(600096:000152) CSIP_sendUpdateMsgFromCh call_id=0->1

neqt=-1->2066 state=NO_SCREEN->SCREEN_DIAL_0_DIGIT

(600096:000153) CSIP_sendUpdateMsgFromCh ->

cnx_create_tab(1, 2066)

(600096:000154) CSIP_constructDistantField UTF-8

SCREEN_DIAL_0_DIGIT key=1

(600096:000155)

""

(600096:000156) CSIP_constructOtherField UTF-8

SCREEN_DIAL_0_DIGIT key=1

(600096:000157)

"PC" 31023

(600096:000158) CSIP_constructSdp Default case

(600096:000159)

172.27.143.131:32584 G_729_A DTMF=101 SIP_SENDRECV

(600096:000160) CSIP_constructOtherInfo clir=0 forward=0

autoAnswer=0

(600096:000161) ..CSIPMsgInFactory::makeMsgInCh

 (600096:000162) ..new CSIPMsgChDial0Digit at 0x54038ce8

- counter 1

(600096:000163) CSIP_sendUpdateMsgFromCh -> call

CSIP_setFeatureList

(600096:000164) nulog_final: 0 typconv : 0 ptdemi-

>forwarded_neqph:-1

(600096:000165) CSIP_setFeatureList

(600096:000168) CSIP_sendInfoCs : No call server

informations authorization..

The Call handling gets the SDP infomation of the

equipment for the RBT to generate the SDP of the

180

(600096:000195)

(600096:000198)

(600096:000199)

(600096:000200)

(600096:000201)

(600096:000203)

(600096:000204)

(600096:000205)

(600096:000206)

(600096:000207)

(600096:000208)

(600096:000209)

(600096:000210)

(600096:000211)

(600096:000212)

(600096:000213)

(600096:000214)

(600096:000215)

(600096:000216)

CSIP_sendInfoCs : No call server informations

authorization.

chgt_local_rtp_info ptdemi->info.hinfo=0 ptdemi-

>neqt=2066

chgt_local_rtp_info local.wc=0 distant.wc=0 before

update

chgt_local_rtp_info end local.wc=0 distant.wc=0

CSIP_sendInfoCs : No call server informations

authorization.

CSIP_sendUpdateMsgFromCh call_id=1->1 neqt=2066->2066

state=SCREEN_DIAL_0_DIGIT->SCREEN_DIAL_DIGIT

CSIP_constructDistantField UTF-8 SCREEN_DIAL_DIGIT key=1

""

CSIP_constructOtherField UTF-8 SCREEN_DIAL_DIGIT key=1

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
" PC" 31023

CSIP_constructSdp Default case

172.27.143.131:32584 G_729_A DTMF=101 SIP_SENDRECV

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
CSIP_constructOtherInfo clir=0 forward=0 autoAnswer=0

..CSIPMsgInFactory::makeMsgInCh

..new CSIPMsgChDialDigit at 0x54038ce8 - counter 1

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CSIP_sendUpdateMsgFromCh -> call CSIP_setFeatureList

nulog_final: 0 typconv : 0 ptdemi->forwarded_neqph:-1

CSIP_setFeatureList

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CSIP_sendInfoCs : No call server informations
authorization.

Here, the IP address for the RBT is 172.27.143.131, and

the port used is 32584 and the codec used is G729

(this information appears few times on the trace)

The 180 is generated by the Call Handling and sent to

the SIPMOTOR.

(600096:000400)

(600096:000401)

(600096:000402)

(600096:000403)

(600096:000404)

Ed. 07

(600096:000405)

(600096:000406)

CSIP_receiveComAction

..activeChId 1 featureList -..CSIP Queue

CSIPMsgChCalledStatus

..CSIPMsgChCalledStatus::getSession

....CSIP_getSessionFromChId

......Retrieved session 1 for ChId 1

..CSIPMsgChCalledStatus::execute

110

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

The “CALL PROC” is present.

The state of the session, for Call Handling point of

view, change to “CSIPStateInvite180WaitConversation”

The Call handling gets the SDP infomation of the

equipment for the 200ok

neqttouc neqt=2049 nekip=2049 toucacod=1

(600121:000476) neqttouc result=1000801 en Hexa !!!

(600121:000477) neqttouc neqt=2049 nekip=2066 toucacod=1

(600121:000478) neqttouc result=1000812 en Hexa !!!

(600121:000479) sip_behind_ice-->neqt=2049,Result=0

(600121:000480) sip_behind_ice-->neqt=2066,Result=0

(600121:000486) SIP ipphone : interro statut 0 ptdemi-

>neqt(2049)

(600121:000487) SIP ipphone : GetneqtEnFace = -1 payload

= 101 neqt =(2066)

(600121:000490) put_rtp_info end 2066 local.wc=0

distant.wc=0

(600121:000497) neqttouc neqt=2066 nekip=2049 toucacod=1

(600121:000498) neqttouc result=1000801 en Hexa !!!

(600121:000499) sip_behind_ice-->neqt=2066,Result=0

(600121:000500) sip_behind_ice-->neqt=2049,Result=0

(600121:000503) numunpack_trace: 31004

(600121:000504) from_same_nb_in_mes :
 nulog=27,numero_lg=5

(600121:000505) CSIP_msg_notify_management : No MWI

subscription.

(600121:000506) sip_behind_ice-->neqt=2066,Result=0

(600121:000507) sip_behind_ice-->neqt=2049,Result=0

(600121:000510) CSIP_sendUpdateMsgFromCh call_id=1->1

neqt=2049->2049
state=SCREEN_CALLED_STATUS>SCREEN_CONVERSATIO

(600121:000511) CSIP_constructDistantField UTF-8

SCREEN_CONVERSATION key=1

(600121:000512)

"IPtouch 172.27.142.64" 31004

(600121:000513) CSIP_constructOtherField UTF-8

SCREEN_CONVERSATION key=1

(600121:000514)

"PC" 31023

(600121:000515) CSIP_constructSdp Default case

(600121:000516)

172.27.142.64:32514 G_729_A DTMF=101 SIP_SENDRECV

(600121:000517) CSIP_constructOtherInfo clir=0 forward=0

autoAnswer=0

(600121:000518) ..CSIPMsgInFactory::makeMsgInCh

(600121:000519) ..new CSIPMsgChConversation at

0x54038ce8 - counter 1

(600121:000520) CSIP_sendUpdateMsgFromCh -> call

CSIP_setFeatureList

(600121:000521) nulog_final: 4 typconv : 1 ptdemi-

>forwarded_neqph:-1

Ed. 07

111

(600121:000522)

CSIP_setFeatureList START_CALL HOLD

(600121:000523) CSIP_sendInfoCs : No call server

informations authorization.

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

Here, the IP address for the 200ok is 172.27.142.64, and

the port used is 32514 and the codec used is

G729, this SDP correponds to the IPtouch.

The 200ok is generated by the Call Handling and sent to

the SIPMOTOR

(600121:000525)

(600121:000526)

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
(600121:000527)

(600121:000528)

(600121:000529)

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
(600121:000530)

(600121:000531)

(600121:000532)

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
(600121:000533)

(600121:000534)

(600121:000535)

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
(600121:000536)

(600121:000537)

(600121:000538)

(600121:000539)

(600121:000540)

(600121:000541)

(600121:000542)

(600121:000543)

(600121:000544)

(600121:000545)

(600121:000546)

(600121:000547)

(600121:000548)

(600121:000549)

(600121:000550)

(600121:000551)

(600121:000552)

(600121:000553)

(600121:000554)

(600121:000555)

Ed. 07

(600121:000556)

(600121:000557)

(600121:000558)

CSIP_receiveComAction

..activeChId 1 featureList START_CALL HOLD

..CSIP Queue CSIPMsgChConversation

..CSIPMsgChConversation::getSession

....CSIP_getSessionFromChId

......Retrieved session 1 for ChId 1

..CSIPMsgChConversation::execute

....CSIPStateInvite180WaitConversation::doCSIPMsgChConversa

......CSIPSession#1ChId#1::setDistantSdp

........CSIPSession#1ChId#1::compareDistantSdp

..........Change 172.27.143.131:32584 G_729_A DTMF=101

SIP_SENDRECV

..........

-> 172.27.142.64:32514 G_729_A DTMF=101 SIP_SENDRECV

........CSIPSession#1ChId#1::resetIsSdpSentInInf

......CSIPSession#1ChId#1::setDistantClir

......CSIPSession#1ChId#1::setDistantName

......CSIPSession#1ChId#1::setDistantNumber

......CSIPSession#1ChId#1::sendSipSuccessful

........CSIPSession#1ChId#1::emitMsgToSIPMotor

..........SIP_SUCCESSFUL sent

+-------------------------------------------------------
-----+

| Message sent UA (neqt : 2066-1) ----> SIP

| Successful 200

 | RELATIVE REQUEST : INVITE

+-------------------------------------------------------
-----+

| SDP :

| @IP:port = 172.27.142.64:32514

| ALGOS :

G729

DTMF : 101

| DIRECTION : SEND & RECEIVE

| AssertedAddress : <IPtouch 172.27.142.64> 31004@oxe-

ov.alcatel.fr:5060

112

| COLP

+-------------------------------------------------------
-----+

......CSIPSession#1ChId#1::changeState

TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

Session Iniation Protcol (SIP)

The state of the session, for Call Handling point of

view, change to “CSIPStateConnectedWaitAck”.

The ACK is received from the SIPMOTOR

(600126:000641) CSIP_receiveSipMsg

(600126:000642) +---------------------------------------
---------------------+

(600126:000643) | Message received SIP ----> UA (neqt :

2066-1)

(600126:000644) | ACK

(600126:000645) +---------------------------------------
---------------------+

(600126:000646) ..activeChId 1 featureList START_CALL

HOLD

(600126:000647) ..CSIPMsgInFactory::makeMsgInSip

(600126:000648) ....SIP_ACK dialogId 1

(600126:000649) ....new CSIPMsgSipAck at 0x54038f90 -

counter 2

(600126:000650) ..CSIP Queue CSIPMsgSipAck <

CSIPMsgChUpdateRtp

(600126:000651) ..CSIPMsgSipAck::getSession

(600126:000652) ....CSIP_getSessionFromId

(600126:000653) ......Retrieved session 1 with ChId 1

(600126:000654) ..CSIPMsgSipAck::execute

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(600126:000655)
....CSIPStateConnectedWaitAck::doCSIPMsgSipAck

(600126:000656) ......CSIPSession#1ChId#1::changeState

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
(600126:000657) ........CSIPStateConnectedWaitAck ->
CSIPStateConnected

The state of the session, for Call Handling point of

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
view, change to “CSIPStateConnected”.

Call released by the OXE:

(600143:000733)

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
(600143:000734)

(600143:000735)

(600143:000736)

(600143:000737)

(600143:000738)

(600143:000739)

(600143:000740)

(600143:000741)

(600143:000742)

(600143:000743)

(600143:000744)

(600143:000745)

(600143:000746)

(600143:000747)

(600143:000748)

(600143:000749)

(600143:000750)

(600143:000751)

The BYE is generated by the Call Handling and sent to

the SIPMOTOR

CSIP_receiveComAction

..activeChId 1 featureList HOLD

..CSIP Queue CSIPMsgChOnHook

..CSIPMsgChOnHook::getSession

....CSIP_getSessionFromChId

......Retrieved session 1 for ChId 1

..CSIPMsgChOnHook::execute

....CSIPStateConnected::doCSIPMsgChOnHook

......CSIPSession#1ChId#1::sendMsgToCh

........CSIP_HANG_UP

......CSIPSession#1ChId#1::sendSipBye

........CSIPSession#1ChId#1::emitMsgToSIPMotor

..........SIP_BYE sent

+-------------------------------------------------------
-----+

| Message sent UA (neqt : 2066-1) ----> SIP

| BYE

+-------------------------------------------------------
-----+

......CSIPSession#1ChId#1::changeState

........CSIPStateConnected -> CSIPStateByeWait200

The state of the session, for Call Handling point of

view, change to “CSIPStateByeWait200”.

(600144:000831)

(600144:000832)

(600144:000833)

(600144:000834)

(600144:000835)

Ed. 07

(600144:000836)

 (600144:000837)

The 200OK of the BYE is received on the Call Handling

CSIP_receiveSipMsg

+-------------------------------------------------------
-----+

| Message received SIP ----> UA (neqt : 2066-1)

| Successful 200

| RELATIVE REQUEST : BYE

113

| No SDP

+-------------------------------------------------------
-----+

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The state of the session, for Call Handling point of

view, change to “CSIPStateIdle”.

The “neqt” is released (SIP_EQT_RELEASED sent)

The “half-com” is released (CSIP lib__demi() called for

neqt 2066)

On the Call Handling, the SIP extension calls have a

“session”, this is the evolution of the session state
from

the INVITE to the 200ok of the BYE:

Ed. 07

CSIPStateIdle -> CSIPStateInviteWaitDial0Digit

o Changing state from the INVITE to the 100 Trying

CSIPStateInviteWaitDial0Digit ->
CSIPStateInviteWaitCalledStatus

o Changing state from the 100 Trying to the 180 Ringing

CSIPStateInviteWaitCalledStatus ->
CSIPStateInvite180WaitConversation

o Changing state from the 180 Ringing to the 200 Ok

CSIPStateInvite180WaitConversation ->
CSIPStateConnectedWaitAck

o Changing state from the 200 Ok to the ACK

CSIPStateConnectedWaitAck -> CSIPStateConnected

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o Changing state from the ACK to the BYE

CSIPStateConnected -> CSIPStateByeWait200

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
o Changing state from the BYE to the 200 Ok of the BYE

CSIPStateByeWait200 -> CSIPStateIdle

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
o Changing state from the 200 Ok of the BYE to the next
INVITE (next call)

114

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TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.10 Main call flows explanation

11.10.1

Forwards

The OXE is able to manage different types of fowards,

when an equipment is doing a forward to a SIP

equipment, according to this forward type, the behavior

and the SIP messages are different.

Topology for explanation:

Legacy phone B (31000)

SIP phone C

(31026)

OmniPCX Enterprise

Legacy phone A (31004)

Ed. 07

115

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.10.1.1

Phone A calls B, and B is in direct foward to C.

In this type of call the OXE is sending an INVITE to C

(for all types of fowards) , here the different type of

INVITEs send according to the declaration of the SIP

equipment on OXE:

C is declared as SIP extension:

----------------------utf8----------------------INVITE
sip:[email protected]:27836;rinstance=e26a48b411982396
SIP/2.0

Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER,

SUBSCRIBE, OPTIONS, UPDATE, INFO

Supported: histinfo,replaces,timer,path

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

Session-Expires: 1800;refresher=uac

Min-SE: 900

Content-Type: application/sdp

To: "IPtouch 172.27.141" <sip:31000@oxe-

ov.alcatel.fr;user=phone>

From: "IPtouch 172.27.142.64" <sip:31004@oxe-

ov.alcatel.fr;user=phone>;tag=fc0ad7be3c9267a849d2

789c08cf26d3

Contact: <sip:[email protected];transport=UDP>

Call-ID: [email protected]

CSeq: 960429378 INVITE

Via: SIP/2.0/UDP
 172.27.141.151;branch=z9hG4bKc2893fd8925d9aa6704859e3fb7887

Max-Forwards: 70

Content-Length: 240

In that case, the important information it is the “TO”

field containing the directory number of the user

forwarded to the SIP extension (31000 in that case), no

mor information to indicate that the call is forwarded.

C is declared as SIP device or an external SIP gateway:

----------------------utf8----------------------INVITE
sip:[email protected]:17680;rinstance=3e53f382fc6e4647
SIP/2.0

Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER,

SUBSCRIBE, OPTIONS, UPDATE, INFO

Supported: histinfo,replaces,timer,path

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

Session-Expires: 1800;refresher=uac

Min-SE: 900

History-Info: <sip:[email protected]?
reason=SIP%3bcause%3d302%3btext%3d%22Moved%20Temporarily%22
<sip:31026@oxe-o

v>;index=1.1

Content-Type: application/sdp

To: <sip:[email protected];user=phone>

From: "IPtouch 172.27.1" <sip:31004@oxe-

ov.alcatel.fr;user=phone>;tag=4200fe39737a85684b86a11b9078a

Contact: <sip:[email protected];transport=UDP>

Call-ID: [email protected]

CSeq: 7963653 INVITE

Via: SIP/2.0/UDP
172.27.141.151;branch=z9hG4bKcbbca67dd61c80b972173fb10c3190

Max-Forwards: 70

Content-Length: 240

v=0

In that case, the important information is the “TO”

field containing the directory number of the user
forwarded

to the SIP extension (31000 in that case), and the field

“History-Info”, this information is present in case of

forward and if it is managed on the OXE side for the SIP

Trunk Group associated to the SIP gateway.

The “History-Info” contains the directory number of the

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
set forwarded, the reason of forward and the

destination of the forward.

The “History-Info” can be changed by “Diversion” for

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
external SIP gateways by management.

Ed. 07

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TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

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TROUBLESHOOTING GUIDE No.0069

The “History-Info” is not validated for SIP extension.

11.10.1.2

Phone A calls C, and C is forwarded to B.

----------------------utf8----------------------SIP/2.0
302 Moved Temporarily

Via: SIP/2.0/UDP
172.27.141.151;branch=z9hG4bK9e0dfb2b8f49bd46aaf944cee38cc4

Contact: <sip:[email protected]>

To: "SIP Phone"<sip:31026@oxe-

ov.alcatel.fr;user=phone>;tag=16325b19

From: "IPtouch 172.27.142.64"<sip:31004@oxe-

ov.alcatel.fr;user=phone>;tag=119145146a704a4541de9

Call-ID: [email protected]

CSeq: 879482083 INVITE

User-Agent: SIP Phone

Content-Length: 0

-------------------------------------------------

Most of the time the SIP equipment returns a 302 message

to inform the proxy that the call is fowarded. This

message is immediat or after a delay according to the

type of forward.

If the SIP equipment is a proxy, it is able to keep the

call, in that case, 2 SIP legs are opened, one from the

OXE to the proxy, the second one from the proxy to the
forwarded destination.

If the SIP equipment is declared as a SIP extension,

from this SIP equipment the forwarding prefixes can be

used, in that case no INVITE will be send to the SIP

equipment, because the Call Handling knows that this

user is forwarded.

11.10.2

Transfer

To make a transfer, the OXE can use (receive and accept)

different ways according to the call context:

The REFER without Replaces

The REFER with Replaces

The REINVITE with Replaces

Topology for explanation:

Legacy phone B (31000)

SIP phone C

(31026)

OmniPCX Enterprise

Legacy phone A (31004)

Ed. 07

117

SIP phone D

(31023)

 TG0069

OmniPCX Enterprise

TROUBLESHOOTING GUIDE No.0069

11.10.2.1

Session Iniation Protcol (SIP)

Use of REFER without replaces.

C calls A and C makes a transfer to B

C sends a REFER to the SIPMOTOR

----------------------utf8----------------------REFER
sip:oxe-ov.alcatel.fr SIP/2.0

Via: SIP/2.0/UDP 172.27.141.210:63016;branch=z9hG4bK-

d87543-5c3865307254f255-1--d87543-;rport

Max-Forwards: 70

Contact: <sip:[email protected]:63016>

To: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=171c87e6f9b80ed5f6819b411a72505c

From: "31026"<sip:[email protected]>;tag=15672359

Call-ID: ODFlNGNmY2JjNDgyOGEwNDRmYjhhY2NjODAxM2U2NWE.

CSeq: 3 REFER

User-Agent: SIP Phone

Refer-To: <sip:[email protected]>

Referred-By: <sip:[email protected]:63016>

Content-Length: 0

-------------------------------------------------

On this REFER, the next information are present:

“Refer-To” contains the the directory number of the

transfer destination.

“Referred-By” contains the directory number of the user

doing the transfer.

The SIPMOTOR sends a 202 Accepted to C

Mon Jun 25 12:04:30 2012 SEND MESSAGE TO NETWORK

(172.27.141.210:63016 [UDP]) (BUFF LEN = 665)

----------------------utf8----------------------SIP/2.0
202 Accepted

Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY, REFER,

SUBSCRIBE, OPTIONS, UPDATE

Contact: sip:oxe-ov.alcatel.fr

Supported: replaces,timer,path,100rel

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

P-Asserted-Identity: "IPtouch 172.27.142.64"

<sip:[email protected];user=phone>

To: "31004" <sip:31004@oxe-

ov.alcatel.fr>;tag=171c87e6f9b80ed5f6819b411a72505c

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
From: "31026" <sip:[email protected]>;tag=15672359

Call-ID: ODFlNGNmY2JjNDgyOGEwNDRmYjhhY2NjODAxM2U2NWE.

CSeq: 3 REFER

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Via: SIP/2.0/UDP
172.27.141.210:63016;received=172.27.141.210;branch=z9hG4bK
d87543-5c386530725

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
4f255-1--d87543-;rport=63016

Content-Length: 0

-------------------------------------------------

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
The 202 Accepted is send to accept the REFER, but the
transfer is not yet done.

The SIPMOTOR sends a NOTIFY to C

----------------------utf8----------------------NOTIFY
sip:[email protected]:63016 SIP/2.0

Content-Type: message/sipfrag

Contact: sip:oxe-ov.alcatel.fr

Supported: replaces,timer,path,100rel

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

Event: refer

Subscription-State: terminated;reason=noresource

To: sip:[email protected];tag=15672359

From: "31004"

<sip:31004@oxe-
ov.alcatel.fr>;tag=171c87e6f9b80ed5f6819b411a72505c

Ed. 07

118

Call-ID: ODFlNGNmY2JjNDgyOGEwNDRmYjhhY2NjODAxM2U2NWE.

CSeq: 1644340323 NOTIFY

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The NOTIFY corresponds to the final state of the

transfer, here the NOTIFY has “200 Ok” at the end of the

message, in that case the transfer has be done by the

OXE.

If the on NOTIFY, the information is 503 Unavailable, in

that case, the transfer has failed. Some other

information can be present (488, 486, etc...) according

to the failed cause.

C replies to this NOTIFY

----------------------utf8----------------------SIP/2.0
200 OK

Via: SIP/2.0/UDP
172.27.141.151;branch=z9hG4bK65961cae897ba970a6b559776cd2cf

Contact: <sip:[email protected]:63016>

To: <sip:[email protected]>;tag=15672359

From: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=171c87e6f9b80ed5f6819b411a72505c

Call-ID: ODFlNGNmY2JjNDgyOGEwNDRmYjhhY2NjODAxM2U2NWE.

CSeq: 1644340323 NOTIFY

User-Agent: SIP Phone

Content-Length: 0

-------------------------------------------------

11.10.2.2

Use of REFER with replaces.

 C calls A and C calls D and makes a transfer

C sends a REFER to the SIPMOTOR to replace an existing

dialog

----------------------utf8----------------------REFER
sip:oxe-ov.alcatel.fr SIP/2.0

Via: SIP/2.0/UDP 172.27.141.210:63016;branch=z9hG4bK-

d87543-d60505761b7d746d-1--d87543-;rport

Max-Forwards: 70

Contact: <sip:[email protected]:63016>

To: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=0219e846e66c868f72a9dbdfa8e58e2a

From: "31026"<sip:[email protected]>;tag=9c131c4f

Call-ID: ZTBjODRhNzFhYTY3ZGNiMjI4N2FjZDQzNTI2MjA2YjE.

CSeq: 7 REFER

User-Agent: SIP Phone

Refer-To: "31023"<sip:[email protected]?
Replaces=YTI4MmJhZjcyMDAyYmYyODI2ZmU0NmE5MWVhMGU2MDc.%3Btot
tag%3D7728f179>

Referred-By: <sip:[email protected]:63016>

Content-Length: 0

-------------------------------------------------

On this call flow there are three legs:

Leg1 corresponds to the call from C to A

Leg2 corresponds to the call from C to D for the

direction C to SIPMOTOR

Leg3 corresponds to the call from C to D for the

direction SIPMOTOR to D

On this REFER, the next information are present:

Ed. 07

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TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

“Refer-To” contains the directory number of the transfer

destination with a “Replaces” corresponding

to the leg to replace (leg2)

“Referred-By” contains the directory number of the user

doing the transfer.

At the end of the transfer the leg1 is closed by C and

leg2 is closed by the SIPMOTOR, it is remaining only

the leg3 from the A to D.

11.10.2.3

Use of REINVITE with replaces.

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
C calls A and C calls D and C makes a transfer

C sends a REINVITE to the SIPMOTOR to replace an

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
existing dialog

----------------------utf8----------------------INVITE
sip:oxe-ov.alcatel.fr SIP/2.0

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Via: SIP/2.0/UDP 172.27.141.210:63016;branch=z9hG4bK-
d87543-71672411fa2ca01c-1--d87543-;rport

Max-Forwards: 70

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Contact: <sip:[email protected]:63016>

To: "31004"<sip:31004@oxe-
ov.alcatel.fr>;tag=0219e846e66c868f72a9dbdfa8e58e2a

From: "31026"<sip:[email protected]>;tag=9c131c4f

Call-ID: ZTBjODRhNzFhYTY3ZGNiMjI4N2FjZDQzNTI2MjA2YjE.

CSeq: 6 INVITE

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY,

MESSAGE, SUBSCRIBE, INFO

Referred-By: <sip:[email protected]:63016>

Replaces=YTI4MmJhZjcyMDAyYmYyODI2ZmU0NmE5MWVhMGU2MDc.%3Bto-
tag%3D053621a0570c23654c20fb10154dd7f5%3Bfromtag%3D7728f179

Content-Type: application/sdp

User-Agent: SIP Phone

Content-Length: 256

The principle is the same than a REFER with replaces,

but it is a REINVITE message

On this REINVITE, the next information are present:

“Referred-By” contains the directory number of the user

doing the transfer.

“Replaces” contains the the directory number of the

transfer destination with a “Replaces”

corresponding to the leg to replace (leg2).

Ed. 07

120

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.10.3

UPDATE on Early Media

In some calls scenarios, the OXE will send or receive an

UPDATE on Early Media (before dialog opened) to

change the SDP.

Topology for explanation:

Legacy phone B (31000)

SIP phone C

(31026)

OmniPCX Enterprise

Legacy phone A (31004)

Phone A calls B, B calls C and makes a blind transfert

to C.

Ed. 07

121

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

During the RINGING phase, the OXE will send to the C a

UPDATE (after sending the 180 RINGING) to C.

 To send the UPDATE, the OXE needs before to send a

PRACK, to make a Pre-Acknowledgment and

receive a 200ok for this PRACK, after this, the OXE will
be able to send an UPDATE.

To send a PRACK the OXE needs a “Require: 100rel” on the

18X answer received:

Mon Jun 11 15:01:38 2012 RECEIVE MESSAGE FROM NETWORK

(172.27.143.186:5060 [UDP])

----------------------utf8----------------------SIP/2.0
180 Ringing

Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY,

SUBSCRIBE, OPTIONS, UPDATE

Contact: sip:172.27.143.186

Require: 100rel

User-Agent: SIP Phone

To:
<sip:[email protected];user=phone>;tag=d7758dbc7f49c9521

From: "IPtouch 172.27.1" <sip:31000@oxe-

ov.alcatel.fr;user=phone>;tag=0c835efa2e1bf86a90d0016a

Call-ID: [email protected]

CSeq: 679245852 INVITE

Via: SIP/2.0/UDP
172.27.141.151;branch=z9hG4bK61c571ebc4b1f5e5ff9e122e7e8b4a

RSeq: 1131790336

Content-Length: 0

-------------------------------------------------

After receiving this “Require: 100rel”, the OXE

generates the PRACK

Mon Jun 11 15:01:38 2012 SEND MESSAGE TO NETWORK

(172.27.143.186:5060 [UDP]) (BUFF LEN = 514)

----------------------utf8----------------------PRACK
sip:172.27.143.186 SIP/2.0

Supported: replaces,timer,path

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

RAck: 1131790336 679245852 INVITE

To:
<sip:[email protected];user=phone>;tag=d7758dbc7f49c9521

From: <sip:31000@oxe-
ov.alcatel.fr;user=phone>;tag=0c835efa2e1bf86a90d0016a0389c

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Call-ID: [email protected]

CSeq: 679245853 PRACK

Via: SIP/2.0/UDP
(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
172.27.141.151;branch=z9hG4bK8b757b21da861556184ff74e5f5aac

Max-Forwards: 70

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Content-Length: 0

-------------------------------------------------

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
The OXE receives the 200ok of the PRACK

Mon Jun 11 15:01:38 2012 RECEIVE MESSAGE FROM NETWORK

(172.27.143.186:5060 [UDP])

----------------------utf8----------------------SIP/2.0
200 OK

Allow: INVITE, ACK, CANCEL, BYE, PRACK, NOTIFY,

SUBSCRIBE, OPTIONS, UPDATE, INFO

Supported: timer,path,100rel

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

To:
<sip:[email protected];user=phone>;tag=d7758dbc7f49c9521

From: <sip:31000@oxe-
ov.alcatel.fr;user=phone>;tag=0c835efa2e1bf86a90d0016a0389c

Call-ID: [email protected]

CSeq: 679245853 PRACK

Via: SIP/2.0/UDP
172.27.141.151;branch=z9hG4bK8b757b21da861556184ff74e5f5aac

Content-Length: 0

-------------------------------------------------

Ed. 07

122

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

The OXE sends the UPDATE to change the SDP.

Mon Jun 11 15:01:38 2012 SEND MESSAGE TO NETWORK

(172.27.143.186:5060 [UDP]) (BUFF LEN = 895)

----------------------utf8----------------------UPDATE
sip:172.27.143.186 SIP/2.0

Supported: replaces,timer,path

User-Agent: OmniPCX Enterprise R10.0 j1.410.45

RAck: 1131790336 679245852 INVITE

To:
<sip:[email protected];user=phone>;tag=d7758dbc7f49c9521

From: <sip:31000@oxe-
ov.alcatel.fr;user=phone>;tag=0c835efa2e1bf86a90d0016a0389c

Call-ID: [email protected]

CSeq: 679245852 UPDATE

Via: SIP/2.0/UDP
172.27.141.151;branch=z9hG4bK8b757b21da861556184ff74e5f5aac

Max-Forwards: 70

Content-Type: application/sdp

 Content-Length: 291

v=0

o=OXE 1339422663 1339422663 IN IP4 172.27.141.151

s=abs

c=IN IP4 172.27.142.64

t=0 0

m=audio 32514 RTP/AVP 18 97

a=sendrecv

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=ptime:20

a=maxptime:40

a=rtpmap:97 telephone-event/8000

-------------------------------------------------

The UAS, receiving this UPDATE, is able to use the

connection point for the RTP flow

11.11 Configuration issues

Most of the SIP issues are linked to a bad management.

When you connect a SIP equipment, it is mandatory to

check if this equipment is tested and validated by

Alcatel-Lucent

The SIP equipments like faxs, sets, etc… are validated

via the AAPP. The

Configuration procedures are available on BPWS.

The SIP providers are tested themselves the OXE, so if

you want to connect one

SIP provider, check if this provider has done the

interopability test. All the

configuration procedures are given by the providers and

not by Alcatel-Lucent.

If a SIP equipment connected is not validated by

Alcatel-Lucent, no support will be provided.

11.11.1

SIP configuration rule

General Parameters

- DPNSS prefix (necessary for optimisation on call

forward).

- System codec (G729, G723).

- Support of multi-algo should be set to false.

Netadmin

- Use of specific characters (& _ $ ...) is not allowed

for the nodename.

Ed. 07

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TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

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TROUBLESHOOTING GUIDE No.0069

Activate internal name resolver in spatial redundancy

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topologies.

Local SIP gateway

- The local SIP gateway is managed when the SIP Trunk

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group and the SIP Subnetwork are

managed (minimum of configuration to do).

Alcatel-Lucent recommends to use an ABCF SIP Trunk Group

on the local SIP

gateway

The network number is a free one, must not used by

another application (ABCF

network, Hybrid links, VPN hop, etc…).

This network number is the same than the one managed on

the SIP ABCF Trunk

Group linked to this local SIP gateway.

External SIP Gateway

- The external SIP gateway can use the same Trunk Group
(TG) than the local SIP gateway.

- The external SIP gateway can use another Trunk Group.

If it is an ABCF TG, the network number manage for this

TG is different than the one

used on the TG for the local SIP gateway.

If it is an ISDN TG, let the OXE manages the network

number by itself, in fact the

configuration is the same than a real ISDN T2/T1.

- If the external SIP gateway is with an ISDN SIP TG,

only ARS must be used, no network or

routing numbers.

- If the external SIP gateway is with an ABCF SIP TG,

network or routing numbers can be

used without restrictions, if the ARS is used, the OXE

must not receive REFER (or

REINVITE with replaces) or 30X messages on this external

SIP gateway (ARS limitation).

SIP Trunk group

- ABCF TG: no restrictions about SIP messages.

- ISDN SIP TG: no REFER (or REINVITE with Replaces) or

30X messages will be sent and

received.

SIP Proxy

- By default, the SIP proxy is set with “SIP Digest” for

the Minimal authentication method, but

there is no Realm managed, so it is necessary to disable

the authentication (SIP None) or to

manage a Realm.

In case of SSH management, the SIP equiments must be

managed as SIP gateway (choice 1).

11.11.2

SIP alarms generated on OXE

On the OXE SIP incidents are generated on Call Handling

side, thes incidents are linked to a SIP alarm (files

under /tmpd), here an example of SIP alarm generated:

Alarm due to Subscriptions:

> 02/07/12 - 15:39:35

 Warning alarm

37F6 [CResponse::checkResponseFields] unknown header is

not applicable for 202/SUBSCRIBE responses

> 02/07/12 - 15:39:35

Minor alarm

[CSubscriptionState::receiveSubscribeMessage] Call:
28844ea68ff53075 eqt: -1 SUBSCRIPTION_STATE failed to

emit a Successful message.

Ed. 07

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In that situation, the OXE receives a “SUBSCRIBE”

message, but is not able to answer it, because the

purpose of this “SUBSCRIBE” message is unknow by the

OXE.

When this type of alarms are present on the OXE, remove

the Subscription on the remote SIP equipment, to

avoid the Alarm.

When lot of alarms like this one are generated on OXE,

they can cause a “crash” of the SIPMOTOR.

Alarm due to bad SIP call context not copied on Stand-By

CPU:

> 02/07/12 - 15:39:35

Warning alarm

37F6 [receiveInviteMessage] StandByCallCreation failed

!.

On the trace, these information are present:

1309553189 ->
[CDuplicateCall::create_duplication_data_struct] _ViaSet
size 218.

1309553189 ->
[CDuplicateCall::create_duplication_data_struct] Via is
bigger than uiCAlcStrStaticGrow:192 RealSize:218.

1309553189 -> ALARM: [receiveInviteMessage]

StandByCallCreation failed !.

In that situation, on the INVITE received, the VIA

header is too long for the OXE and the it is not able to
send

the SIP “context” to the stand by CPU. The call is

established, but in case of bascul, this will not be
known by

the new main CPU.

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Alarm to emit an INVITE message:

> 02/07/12 - 15:39:35

Minor alarm

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[receiveInviteEvent] Call: eqt: 30311 INITIAL_STATE
failed to emit an Invite message.

When the Information is “receiveInviteEvent”, the Call

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Handling is sending an INVITE to the SIPMOTOR, but

due to a lack of ressources or licenses the INVITE

cannot be emitted by the SIPMOTOR.

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> 02/07/12 - 15:39:35

Minor alarm

[receiveInviteMessage] failed to emit an Invite event.

When the Information is “receiveInviteMessage”, the

SIPMOTOR has recieved an INVITE but due to a lack

of ressources (channels on SIP Trunk Group, CAC,

compressors, ...) or licenses, the SIPMOTOR cannot

send the INVITE to the Call Handling.

Alarm due to a request not for the SIP proxy of the OXE:

> 06/05/12 - 21:56:44

Warning alarm

[CIOCom::receiveResponse] Received response is not for

this entity

This alarm means that the SIPMOTOR receives a SIP

resuqets not for it, and is not able to rout it to
another

SIP equipment. Necessary to make a SIPMOTOR traces to

get the IP address of this SIP equipment.

Alarm to emit a SIP message MESSAGE:

> 06/05/12 - 22:14:46

Minor alarm

[receiveMessageEvent] Call: eqt: 2862 INITIAL_STATE

failed to emit an instant message.

The SIPMOTOR is not able to send a SIP message MESSAGE

to a SIP extension, remove the fact to send

this message on the SIP extension phone cos.

Alarm to emit a SIP message CANCEL:

Ed. 07

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TROUBLESHOOTING GUIDE No.0069

> 03/08/12 - 09:31:11

Minor alarm

[receiveCancelEvent] Call:
[email protected] eqt:
2175 COMPLETED_STATE

failed to emit a Cancel message.

The SIPMOTOR generates this alarm because it is not able

to send a CANCEL message, because the

dialog is already opened. The Call Handling asks the

SIPMOTOR to send a CANCEL, but the 200ok for this

INVITE transaction is already arrived.

Alarm to emit a SIP message ACK:

> 02/24/12 - 16:31:42

Minor alarm

[receiveAckEvent] Call:
[email protected] eqt:
2175 TERMINATED_STATE failed

 to emit an Ack message.

The SIPMOTOR generates this alarm because it is not able

to ACK an INVITE transaction, because the

transaction is already terminated. Need to open a SR for

analyse.

11.11.3

Common SIP issues

This part is used to explain the general issues possible

on the OXE, not for a specific equipment

SIPMOTOR

11.11.3.1

Issue 1:

Symptom: With the command ps -edf | grep sipmotor, no

processes are present

Explanation: This is due to a bad configuration of the

SIP on your OXE, for instance, the SIP

Trunk group manage on the local SIP gateway is not a SIP

Trunk Group.

Solution: Manage the good configuration, and after a

restart of the CPU is mandatory.

Issue 2:

Ed. 07

No SIPMOTOR processes are running

Only 2 SIPMOTOR processes are running

Symptom: With the command ps -edf | grep sipmotor, only

2 SIPMOTOR processes are

present

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TROUBLESHOOTING GUIDE No.0069

Explanation: When a modification is done on the SIP

Trunk Group associated to the local

SIP gateway, for instance to replace Mini SIP Trunk

group by a SIP Trunk group, the OXE

needs do resize the memory space due to this

modification (often after the first management

of the local SIP gateway)

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Solution: A restart of the CPU is mandatory

Issue 3:

SIPMOTOR in degraded mode

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-

Symptom: SIPMOTOR is rejecting all the call by a 503

message, and with the tool

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“sipdump”, the status of the SIPMOTOR is in “degraded”
mode

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Explanation: This a protection for the SIPMOTOR, when
there are too much SIP “instance”

in the SIPMOTOR, the SIPMOTOR switches in degraded mode

to protect itself. When it has

this status, all the incoming SIP requests are rejected

by a 503. This mechanism avoids the

application from being overwhelmed by the traffic.

Solution: nothing can be done, the SIPMOTOR will disable

this mode automaticaly due to

some internal timers and thresholds.

Issue 4:

Losing all the SIP call contexts

Symptom: If a restart of the SIPMOTOR is performed, all

the SIP call contexts are lost

Explanation: The restart of the SIPMOTOR provides the

loss of all the SIP contexts, that

means, if you have SIP calls established, the RTP flow

is maintained, for the SIP point view

the call is n ot anymore present, that means, if the

SIPMOTOR receives a BYE for a call, the

BYE will be answered by a “481 Call/Transaction Does Not

Exist”, but the call will be

stopped. Also if you use the session timer (time to

check if the call is still up for the SIP point

of view) the call will be cut by the OXE because, the

context is unknown by the SIPMOTOR

Solution: This is a normal behaviour if the restart is

done manually. If the SIPMOTOR restart

automaticaly, in case a SR must be opened for analyse.

Issue 5:

SIPMOTOR memory leak.

Symptom: The SIPMOTOR is using more and more memory

space.

Explanation: When the SIP is managed on the OXE, the

SIPMOTOR processes are using

memory space, when the traffic is going up, the memory

space used is increasing, when the

traffic is going down, the memory space used is

decreasing. Now, when the traffic is going

down, the memory space used doesn‟t decrease correctly,

and if day after day, when there

is no traffic, the memory used is more higher, the

SIPMOTOR will crash soon. In such case,

the SIPMOTOR has problems to “delete” the SIP contexts

 from its memory, and after

accumullation of the SIP contexts not deleted, the

SIPMOTOR cannot work properly, and

crash.

Action to do:

Ed. 07

Check if the configuration of the OXE respects the

Alcatel-Lucent

recommendations.

Check if the REGISTER messages received on SIPMOTOR are

not too much,

the registration of a SIP equipments must not be used as

a “keep alive”.

Check if the SIPMOTOR doesn‟t receive SIP messages not

for it.

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Solution: A restart of the SIPMOTOR can be done and due

to this, all the SIP contexts are

deleted. The problem will be solved but only for a time,

if the root cause is not found, the

problem will be back again. Open a SR for analyse.

Call failure

11.11.3.2

Issue 1:

Ed. 07

Check if the SIPMOTOR doesn‟t receive SUBSCRIBE messages

not used by

OXE.

Incoming SIP calls are cut by the OXE after 32 seconds:

Symptom: Incoming SIP calls are cut by the OXE after ~3

secondes (or 32 seconds in case

of SIP extension) and the 200ok from OXE is never ACK by

the external SIP equipment.

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Explanation: If the system is in spatial redundancy,
check if the FQDN of the OXE is used by

the external SIP equipement, in fact on the “Contact”,

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the FQDN is added by the OXE, this

FQDN is unknown by the SIP equipment (because it is

using the IP address), and it doesn‟t

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answer to this 200ok, the OXE send several times the
200ok and the OXE cuts the call

because no ACK for this call.

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-

Solution: The remote SIP equipment must use the FQDN of

the OXE. Since the R10, a

parameter is present on the external SIP gateway only

“Contact with IP address” used to put

the IP address of the main CPU instead of the FQDN in

the Contact header.

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Issue 2:

Symptom: The SIP calls are not possible thru an external

SIP gateway in high traffic.

Explanation: Check if the IP address managed on the

external SIP gateway is put in

quarantine (in sipalarm files)

Solution: Manage the IP address on the trusted SIP IP

addresses. A restart of the

SIPMOTOR is mandatory after management.

Issue 3:

SIP calls are rejected with a 502:

Symptom: When a SIP call, using an ABCF SIP Trunk Group,

to an external number is not

possible (thru a carrier for instance) and rejected most

of the time by a 502 Bad Gateway.

Internal calls are ok and incoming calls also ok for

this SIP equipment.

Explanation: When the message 502 is reponded to a SIP

request, the problem is due to the

management, that means, the information on the SIP

request are not good for the call in

progress. In that case, the call is done from an ABCF

SIP Trunk Group to an external called

party, the call is rejected because the DID transcoding

is set to “True” on the ABCF SIP

Trunk Group

Solution: Set the “DID transcoding” of the SIP ABCF

Trunk group to false (mandatory).

Issue 4:

SIP calls are rejected with a 488 Not Acceptable here:

Symptom: When a SIP call is rejected by 488 SIP message,

Explanation: When a SIP call arrives on the OXE, the

Call Handling checks if the SDP

received is compatible for this call, if it is not the

case, the Call Handling asks the

SIPMOTOR to send a response 488 for this request

Solution: Manage the SDP of the SIP equipment to be

compatible with the configuration of

the OXE or on the other way

Issue 5:

SIP calls are rejected with different reasons:

Symptom: When a SIP call is rejected by 488, 502, 404,

etc...

Explanation: When a SIP call arrives on the OXE, this

call is automatically rejected by OXE,

but the reason can be different, even if the scenario of

the call is the same. The SIP is linked

to the shelf 19 associated to the CPUs, so if the CPUs

are not belonging to the IP domain 0,

the virtual INTIP boards of the shelf 19 doesn‟t belong

to the IP domain 0, and the SIP is

affected by this configuration.

Solution: Manage CPUs IP addresses on the IP domain 0,

this mandatory in case of SIP.

Issue 6:

Ed. 07

Calls are not anymore possible from a SIP equipment:

SIP calls are rejected with 403 No license available:

Symptom: When a SIP call is rejected by 403 No license

available.

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TROUBLESHOOTING GUIDE No.0069

Explanation: When a SIP call is done, a license is used

for this call. In case of incoming call,

if no more license is available, the OXE rejects the

call by a 403 No licenses available. The

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problem can be only the number bought by the customer,
it is no enough according to the

number of simultaneous SIP calls, or some SIP call

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contexts are blocked on the

SIPMOTOR.

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Action to do:

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

When no more SIP calls, restart the SIPMOTOR.

Run the SIPMOTOR traces:

>motortrace 3 (or 6)

>traced -l /tmpd/traced -s 10000000 -f 50 -d &

Keep the trace running until the issue is present.

When the issue is present, run “sipdump” and make the

choice 1 and 4 every

minutes during 5/10 minutes.

Stop the traces

When no more SIP calls are present on OXE, run the next
trace (do not restart

the SIPMOTOR!!!):

>motortrace 3 (or 6)

>traced >/tmpd/trace_sip.log and make one call and stop

it.

On the file “trace_sip.log”, search for “nb available

licenses=”.

11.11.4

Solution: If the number of licenses is the number of the

licenses bought on OXE, there is no

issue, the solution is to buy more licenses. If the

number is less than the number bought,

open a SR and provide the traces files and the

Infocollect of the site.

SIP Device issues

An important thing to remenber about SIP device, it is

that all the calls are linked to the SIP Trunk Group

associated to the local SIP Gateway, so if you manage a

SIP ABCF Trunk Group or an ISDN SIP Trunk

Group, the behaviour will be different.

Issue 1:

Ed. 07

Forward on no reply doesn‟t work when the destination is

a SIP device:

Symptom: It is not possible to make a forward on no

reply (on an IPtouch for instance) when

the destination is a SIP device, ok for immediat foward.

Explanation: The SIP device behavior is linked to the

SIP Trunk group associated to the

local SIP gateway, if you use an ISDN SIP TG, or an ABCF

SIP TG, the behaviour will be

different. The SIP Trunk Group used on the local SIP

gateway is a SIP ISDN TG.

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Issue 2:

Solution: Change the SIP Trung Group managed on the

local SIP gateway from SIP ISDN

TG to SIP ABCF TG, restart of the SIPMOTOR is mandatory.

Afer a while, all SIP phones registrations and

subscriptions are impossible

Symptom: More 1000 SIP Devices losses their

registration. Only a double bascul of PBX

resolves this issue

Explanation: As there are more 1000 SIP devices which

register/subscribe at the same

moment, there are too much of traffic to be managed by

the PBX and resources on

SIPMOTOR are blocked. Around 45000 Subscription and

Registration been handled in 3

hours time. This is really a big number,Oxe is dealing

with. Solution shoud be to stop some

of the unwanted Subscribe messages. And increase the

subscriptions and registration

timers on SIP Devices. Unwanted subscriptions meant here

was even though voice mail was

not configured for a phone set, subscription value was

configured, this should be 0.

Example of Registration too brief:

Sun Sep 30 06:53:09 2012 RECEIVE MESSAGE FROM NETWORK

(172.30.125.75:5060 [UDP])

----------------------utf8----------------------REGISTER
sip:172.30.127.2:5060 SIP/2.0

Expires: 60

1348980789 -> Sun Sep 30 06:53:09 2012 SEND MESSAGE TO

NETWORK (172.30.125.75:5060 [UDP])

(BUFF LEN = 394)

----------------------utf8----------------------SIP/2.0
423 Registration Too Brief

Min-Expires: 1800

Example of sipalarm when subscription is impossible on

/tmpd:

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[CSubscriptionState::receiveSubscribeMessage] eqt: -1
SUBSCRIPTION_STATE failed to emit a Successful message.

Example of DHCP buffer issue on /varlog/messages:

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Nov 7 00:01:52 sr_cpub dhcpd: send_packet: No buffer
space available

Nov 7 00:01:52 sr_cpub kernel: Neighbour table overflow.

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Nov 7 00:01:52 sr_cpub kernel: Neighbour table overflow.

Solutions:

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1. Increase registration and susbcriptions timers on SIP
Devices from 60 secondes to

1800.

2. Deactivate unnecessaries subscriptions sent to PBX

when no services are configured

on users management, example: if Voicemail is available

via another application,

subscription must not sent to PBX

3. Configure a dedicated VLAN for OXE (CS, GD) and one

or more VLANs for SIP

Devices in order to decrease ARP requests on DHCP

service

With the current Linux OS, OXE has a limitation in

handling more than 1000 data equipment if it

is connected in the same sub-network. So we need to have

a seperate VLAN in between to

handle this. OXE CS must be placed under separate subnet

and the IP Phones distributed under

different other subnets

11.11.5

SIP extension issues

The SIP extension is not linked to a SIP Trunk Group, it

can be created without SIP management

Issue 1:

Ed. 07

SIP fax equipment, declared as a SIP extension, doesn‟t

work:

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Symptom: when a SIP fax equipment tries to make a call,

the REINVITE for the T38

negociation is never seen

Explanation: When a SIP fax call is done, the

establishement of the call is done in two

phases, opening of RTP channel then opening of a T38

channel, in case of SIP extension,

the T38 is not implemented, so the second phase cannot

be done, and the call is stopped

Solution: Use of a SIP Device user instead of a SIP

extension

Issue 2:

Symptom: when a SIP extension is created, it is a

multiline user, and if the SIP phone is

 associated is monoline, the functioning of the SIP

extension can cause issue

Explanation: A SIP extension user, declared in

“business” mode, is multiline, that means taht

teh SIP phone associated must be multiline as well, if

it is not the case, the call to the

second line of the user is rejected by the SIP phone,

and this can cause disturbances on the

SIP extension behaviour (call handling side) .

Solution: A SIP phone associated to a SIP extension user

must be multiline.

11.11.6

Issue 1:

SIP External Gateway Issue

One way calls after remote SIP equipment put on hold and
retreive the call:

Symptom: A SIP call is done between the OXE and a remote

SIP gateway, this SIP

equipment puts on hold the call, the OXE equipment can

hear the MOH, and when the SIP

equiment retrieves it, the one way call is present.

Explanation: When the SIP external gateway put on hold,

it is sending a REINVITE with a

“Black Hole” (c=0.0.0.0 on SDP) or an “INACTIVE” to stop

the RTP flow, before to send a

new REINVITE with a SDP for MOH. When a new REINVITE is

sent to get back the

converstaion, the OXE is not able to connect the RTP

flow to the SDP given on this

REINVITE.

Solution: On the external SIP gateway, put True for the

parameter “Ignore inactive/black

hole”, in that case, the OXE will not take into account
the “Black Hole” or the “INACTIVE”.

Issue 2:

Ed. 07

SIP extension multiline, SIP phone monoline:

One way call in case of incoming/outgoing calls:

Symptom: An incoming or an outgoing calls are well

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
established, but no speech send by

OXE

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Explanation: The problem has been seen after an upgrade
from a version lower to I160516c

to a R10. On the traces taken, the OXE is not getting

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
SDP or, INVITE or 200ok. The problem

was about the parameter “Routing Application”, this

parameter is used for the feature

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
“Force_on_NET”, in case of incoming call to the OXE,
this call is not for an equipment

connected to the OXE, but for an external user (mobile

phone for instance), so for such call,

the OXE doesn‟t need to reserve ressources on its side.

This parameter has been designed

for that.

Solution: Put False for this parameter if it set to

True.

132

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.12 Use case

11.12.1

Outgoing Call – Cancel sent by OXE after 180 w SDP

Symptom: SIP ISDN Outgoing call are cancelled by OXE

after 180 Ringing SDP (G711) reception.

Diagnosis: - Check if CS‟s IP Address is configured on

IP Domain 0.

- Check extra domain codec where caller is located

Solution: As only G711 codec is available for Outgoing

calls ( IP Compression Type + G711 on TG) and

caller is located on a restricted domain (Extra Domain

Coding Algorithm + With Compression on IP

Domain), OXE cannot sends/receives media stream. Call is

cancelled.

11.12.2

Telephone-event are not provided on SDP offer

Symptom: Re-INVITE sent by OXE to SIP Provider doesn‟t

contain telephone event media on SDP offer

Solution: On SIP > SIP External Gateway, check at false

parameter “To EMS”.

11.12.3

Loss of communication with SIP External Voicemail

Symptom: Frequent loss of communication between external

voicemail and OXE connected via SÏP trunk

Diagnosis: Check if congestion occurs with incident 5816

when you try to access to the voice mail.

Check if Voicemail IP Address is present on Trusted IP

Addresses

Solution: Voicemail was put in quarantine and during one

half hour all calls in direction of Voicemail were

blocked

11.12.4

Impossible to let a message when routing via SIP

Automated Attendant

Symptom: It is not possible to let a message on the

 voicemail of the called number in case of an automated

attendant SIP and when the Phone Feature COS “Voicemail

forwarding” is set at “Ring called set mail”

Solution: On System > Other System Param. > Spec.

Customer Features Parameters > Voice Mail

forwarding SIP auto att, check this parameter at true

11.12.5

When call is transfer from a Third Party Server, after

few seconds, a Re-Invite is

sent by OXE to reroute RTP to a GD card

Symptom: When call is established, after few seconds,

OXE sends a reinvite request to redirect RTP to a GD

card.

Solution: DPNSS is used on this scenario. On System >

Other System Param. > External Signalling

Parameters > DeActivate Path Replacement, check this

parameter at true

11.12.6

Incoming call from a SIP Third Party Server is rejected

by OXE with a SIP Error 488

Not Acceptable Here

Symptom: Incoming call is rejected by a SIP Error 488

Not acceptable Here

Diagnosis: Check Extra Domain Coding Algorithm

concordance

Check Public Access Category

Ed. 07

133

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Solution:

On IP > IP Domain > Extra Domain Coding Algorithm must

be the same as third party offer

On Categories > Access Category > Go down hierarchy >

Public Access Category > Select COS 31

and give correct rights

11.12.7

Incoming call is not recognized as INTERNATIONAL

Symptom: Incoming call received on set phone indicates

local call instead of international call.

Diagnosis: - Country code is not separated of received

number by PBX so canonical form is not correctly

set up. Canonical form is “+” country code “–” *

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
(number). So, number should be +49–71182137821 in

order to detect that is an international incoming call.

Solution: Add the country code 49 on External Country

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Code section Translator > External Numbering Plan >

Coutry Codes:

Country code prefix : 49

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
Country Value + Germany

11.12.8

When we attempt to register on SIP External Gateway, OXE

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
answers by a SIP error

“482 Loop Detected”

Symptom: For each register sent to OXE, we have a SIP

error “482 Loop Detected”, as below REGISTER

request:

1352974529 -> Thu Nov 15 11:15:28 2012 SEND MESSAGE TO

NETWORK (172.27.139.90:5060 [UDP]) (BUFF LEN = 478)

----------------------utf8----------------------REGISTER
sip:hq2cs.labjtr.fr SIP/2.0

Supported: 100rel,path

User-Agent: OmniPCX Enterprise R10.1 j2.501.16.c

To: sip:[email protected]

From:
sip:[email protected];tag=a9ca34e0b0534fb9d4e0823b7b5d4e

Contact:
<sip:[email protected];transport=UDP>;expires=1800

And error received:

Thu Nov 15 11:15:28 2012 RECEIVE MESSAGE FROM NETWORK

(172.27.139.90:5060 [UDP])

----------------------utf8----------------------SIP/2.0
482 Loop Detected

To: sip:[email protected]

From:
sip:[email protected];tag=a9ca34e0b0534fb9d4e0823b7b5d4e

Call-ID: [email protected]

CSeq: 1821162596 REGISTER

Via: SIP/2.0/UDP
172.27.145.122;branch=z9hG4bK47b7d67d20268bb0c40d57c60e4c1c

Content-Length: 0

Diagnosis: Registration is done by Domain Name

resolution so the sip Request-URI sip:hq2cs.labjtr.fr
must

be matched with machin name filled on SIP Gateway. The

SIP URL of REGISTER contains the SRV/A

domain name. Proxy loops that call back to itself

because it does not know about itself as the SRV/A
domain.

Solution: Modify the SIP Gateway in order to have the

same Machin Name as SIP URL contained on

REGISTER, use the command netadmin to do it:

Trunk Group : 35

IP Address : 172.27.139.90

Machin name : hq2cs.labjtr.fr

Proxy Port Number : 5060

DNS local domain name : labjtr.fr

DNS type + DNS A

First DNS IP Address : 172.27.139.88

 Ed. 07

134

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.12.9

When we attempt to register our SIP External Gateway

with an external SIP Proxy,

SIP Proxy answers by a SIP error “416 Unsupported URI

Scheme”

Symptom: For each register sent to external SIP Proxy,

we have a SIP error “416 Unsupported URI

Scheme”, as below REGISTER request:

1352975879 -> Thu Nov 15 11:37:56 2012 RECEIVE MESSAGE

FROM NETWORK (172.27.145.122:5060 [UDP])

----------------------utf8----------------------REGISTER
sip:hq2.labjtr.fr SIP/2.0

Supported: 100rel,path

User-Agent: OmniPCX Enterprise R10.1 j2.501.16.c

To: sip:hq2.labjtr.fr

From:
sip:hq2.labjtr.fr;tag=56b8ce5bd76524902b5c171f39c9bbdf

Contact: <sip:172.27.145.122;transport=UDP>;expires=1800

Call-ID: [email protected]

CSeq: 1643105352 REGISTER

Via: SIP/2.0/UDP
172.27.145.122;branch=z9hG4bKdc224f76827da20ba9390b081ef8ae

Max-Forwards: 70

Content-Length: 0

And error received:

Thu Nov 15 11:37:56 2012 SEND MESSAGE TO NETWORK

(172.27.145.122:5060 [UDP]) (BUFF LEN = 344)

----------------------utf8----------------------SIP/2.0
416 Unsupported URI Scheme

To:
sip:hq2.labjtr.fr;tag=75e766ee37e6bf967b4c84db521f8406

From:
sip:hq2.labjtr.fr;tag=56b8ce5bd76524902b5c171f39c9bbdf

Call-ID: [email protected]

CSeq: 1643105352 REGISTER

Via: SIP/2.0/UDP
172.27.145.122;branch=z9hG4bKdc224f76827da20ba9390b081ef8ae

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Content-Length: 0

Diagnosis: Registration ID is not present on REGISTER

request so SIP Proxy cannot authenticate the OXE.

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Configure the parameter Registration Id on SIP External
Gateway

Solution: Configure the parameter Registration Id on SIP

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
External Gateway, as well

1352976351 -> Thu Nov 15 11:45:50 2012 RECEIVE MESSAGE

FROM NETWORK (172.27.145.122:5060 [UDP])

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
----------------------utf8----------------------REGISTER
sip:hq2.labjtr.fr SIP/2.0

Supported: 100rel,path

User-Agent: OmniPCX Enterprise R10.1 j2.501.16.c

To: sip:[email protected]

From:
sip:[email protected];tag=bfc35e619db3ff4f042097e7b390c30a

Contact:
<sip:[email protected];transport=UDP>;expires=1800

Call-ID: [email protected]

CSeq: 571892426 REGISTER

Via: SIP/2.0/UDP
172.27.145.122;branch=z9hG4bK8d42eea8f1c72df626c86ea191f7ff

Max-Forwards: 70

Content-Length: 0

Thu Nov 15 11:45:50 2012 SEND MESSAGE TO NETWORK

(172.27.145.122:5060 [UDP]) (BUFF LEN = 396)

----------------------utf8----------------------SIP/2.0
200 OK

Contact:
<sip:[email protected];transport=UDP>;expires=1800

To:
sip:[email protected];tag=2810b4ed27aa41ba89b99ef3631a8c0d

From:
sip:[email protected];tag=bfc35e619db3ff4f042097e7b390c30a

Call-ID: [email protected]

CSeq: 571892426 REGISTER

Via: SIP/2.0/UDP
172.27.145.122;branch=z9hG4bK8d42eea8f1c72df626c86ea191f7ff

Content-Length: 0

11.12.10

Incoming call doesn’t transit via Trunk Group configured

on SIP Ext Gw

Symptom: When we make a trkvisu of SIP Trunk Group used

by SIP External Gateway during an incoming

call, we observed that no SIP Access is used.

Ed. 07

135

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Diagnosis: - by checking INVITE request received from

Network, we can see that domain contained on

FROM header is not recognized by SIP External Gateway,

 so call transits through Main SIP Gateway.

1332292333 -> Wed Mar 21 02:12:13 2012 RECEIVE MESSAGE

FROM NETWORK (172.27.138.36:5060 [UDP])

----------------------utf8----------------------INVITE
sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP
172.27.138.36:5060;branch=z9hG4bK15ac35dc;rport

Max-Forwards: 70

From: "Boss Hoggs"

<sip:[email protected]>;tag=as5ff02451

To: <sip:[email protected]>

Wed Mar 21 02:12:13 2012 [isDomainFromGwExt] Host from

request is : 172.27.144.20.

Wed Mar 21 02:12:13 2012 [isDomainFromGwExt] User from

request is : 0033XXXXXXXXX

Wed Mar 21 02:12:13 2012 [domain not from an External

Gateway.

Wed Mar 21 02:12:13 2012

11cd[CMotorCall::onReceiveRequest] system option=0
extGw=-1.

Wed Mar 21 02:12:13 2012

11cd[CMotorCall::toGatewayOrProxy] request for
proxydomain=172.27.144.20.

Solution: Modify FROM header sent by external

application in order to match with remote domain
configured

on SIP External Gateway

11.12.11

Wrong caller number sent in case of forward

Symptom: Wrong caller number on OpenTouch anymobile

device when using multi device feature.

Example:

External user 0980406562 (phone A)

OT MIC SIP directory number 7905 (358306667905) (phone

B)

OT anymobile number +358 (0) 505307947 (phone C)

Phone A calls phone B with a redirection to phone C.

During phone C ringing phase, Calling Number

of phone B is displayed instead of Calling number of

phone A

Diagnosis: - Check if history-info/diversion header is

present on requests received from OpenTouch with

related forward informations

- Check External Signalling Parameters (Calling Name

Presentation, NPD for external forward

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
Solution: NPD for external forward is configured at -1
so OXE sends redirecting number in case of forward.

When parameters is configured with NPD used by SIP Trunk

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Group, initial Calling Number is sent.

Before NPD modification:

P-Asserted-Identity: "0501636"
(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
<sip:[email protected];user=phone>

Content-Type: application/sdp

To: <sip:[email protected];user=phone>

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
From: "0501636"
<sip:[email protected];user=phone>;tag=77b6c14021

Contact: <sip:[email protected];transport=UDP>

After NPD modification:

P-Asserted-Identity: "0501636"
<sip:[email protected];user=phone>

Content-Type: application/sdp

To: <sip:[email protected];user=phone>

From: "0501636"
<sip:[email protected];user=phone>;tag=10067c3f78682c28

Contact: <sip:[email protected];transport=UDP>

11.12.12

Diversion/History-Info header is not present

Symptom: User A (+33298285305) calls user B (1481001)

located on PBX. User B is on immediate forward

to User C (+33675445566). Second leg transits via the

Trunk Group 16 (SIP ISDN All Countries) and SIP

External Gw 2 (Remote domain: 172.44.266.44). Diversion

header is not added by OXE.

Diagnosis: - Check External Signalling Parameters, Trunk

Group and SIP External Gateway configuration

Solution: Configure following parameters:

System > Other System Param > External Signalling

Parameters

Ed. 07

136

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

NPD for external forward: 10 (NPD used by SIP ISDN Trunk

Group)

Trunk Groups > Trunk Group

IE External Forward: Diverting leg information

SIP > SIP Ext GW

Diversion Info to provide via: Diversion

(013064:000323) | Diversion :

(013064:000324) |

Url : <> [email protected]

(013064:000325) |

Reason : UNCONDITIONAL

11.12.13

SIP-Trunking Name is displayed on calling phone set when

call is established

Symptom: SIP Trunking Name is displayed on calling phone

set when call is established with an external

user through SIP Externl Gateway. SIP Trunk type is ISDN

ALL COUNTRIES. Example: A is an internal

 phone set and dials external number +33014596222, when

call is established, phone set doesn‟t display

called number

Diagnosis: Check if SIP Carrier sends a P-Asserted-

Identity header on SIP 200 OK Response when call is

established.

Solution: Symptom: If no Called information is present

on connection message (SIP 200 OK), OXE by default

displays the trunk group name.

11.12.14

From header has not the national format

Symptom: Bad tagging of the calling from a SIP ISDN

gateway

Diagnosis: When value on From header is not canonical,

OXE tags the calling number like RNIS unknown

Solution: Modify the from received on OXE by adding

canonical form and manage the country code like this

the calling number will be tagged as national

11.12.15

Incoming and outgoing fax communications impossible

through SIP Gw

Symptom: Re-INVITE with T38 on SDP is not sent by FAX

Server, voice communication is cut before T38

négotiation

Diagnosis: As PBX is configured in spatial redundancy,

FQDN is used. In this case, FQDN corresponds to

the nodename concatenate with the DNS local domain name

managed on SIP Gw. When OXE makes a fax

call to Fax Server, FQDN is used on CONTACT header and

as Fax Server cannot resolve it, call is cut.

Solution: Use an external DNS server for FQDN resolution

or check at false the “Contact with IP Address”

parameter on SIP Ext Gw.

11.12.16

No Re-Invite with T38 offer sent by OXE

Symptom: No T38 bascul during fax communication between

PBX and FAX Gw

Diagnosis: On INVITE sent by the FAX Gw, FROM header

contains the IP Address of PBX instead of IP

Address of FAX Gw. So, when a Fax call arrives, this is

the internal Sip Gw on PBX that is used and SIPABCF
trunk group associated. RE-INVITE(T38) is only available
on trunk group SIP ISDN.

Solution: Modify the IP Address on From Header sent by

Fax Gw

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
11.12.17

External call with secret identity over SIP Provider

fails

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Symptom: Impossible to receive incoming calls with the
secret ID

Ed. 07

(https://pinterest.com/pin/create/button/?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroublesho
137

TG0069

OmniPCX Enterprise

(https://www.reddit.com/submit?url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftroubleshooting-gu
Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

Diagnosis: When a call is received with the secret ID,

the call is rejected by OXE with a 480 (not able to

reach the third party)

Solution: The OXE is using the FROM field for the SIP
gateway selection, in case of secret id, the FROM

field contains this: [email protected], so an

external SIP gateway should correspond to the

domain part of the URI, in that case anonymous.invalid

(SIP Remote domain), this external SIP gateway has

the same configuration than the one used to reach the

SIP provider.

11.12.18

On SIP outgoing call, dynamic ports are used instead of

port 5060

Symptom: why the OXE uses one of the dynamic ports for a
SIP call instead of the port 5060?

Diagnosis: When a SIP trace is done with “wireshark”,

the source port, when the OXE is the initiator of the

call, can be different than 5060 (SIP port manage on the

database)

Solution: Regarding the RFC3581, the initiator of the

SIP call can choose a port number different than the

default “SIP port” (5060) for its source port. So in

that case the OXE is able to choose one port from the

range of dynamic ports.

The important impacts about this behavior, it is the

management of the size of dynamic ports range and also

to take into accounts the configuration of the firewalls

from the customer„s network, to authorize them to use

the dynamic ports for SIP communication.

11.12.19

A "+" character is added on calling number when ISDN

call is routed to SIP

Diagnosis: Addition of "+" is normal, because incoming

call from ISDN is tagged with 21 81 which

corresponds to a National Call and according to the RFC,

a “+” must be added before the Calling Number

___________________________________________________________

| (033539:000002) Concatenated-Physical-Event :

| long: 40 desti: 0 source: 0 cryst: 1 cpl: 6 us: 0

term: 0 type a5

| tei: 0 >>>> message received : SETUP [05] Call ref :

00 37

|__________________________________________________________

| IE:[04] BEARER_CAPABILITY (l=3) 80 90 a3

 | IE:[18] CHANNEL (l=3) a9 83 8c -> T2 : B channel 12

exclusive

| IE:[6c] CALLING_NUMBER (l=6) -> 21 81 Num : 2000

| IE:[7d] HLC (l=2) 91 81

|__________________________________________________________

Solution: The "+" is added because the calling party is

tagged "national" on the ISDN call, so the OXE ia

added the "+". None configuration must be done on OXE

side.

11.12.20

Diversion Field has not the canonical form

Symptom: User A (+33298285305) calls user B (1481001)

located on PBX. User B is on immediate forward

to User C (+33675445566). Second leg transits via the

Trunk Group 16 (SIP ISDN All Countries) and SIP

External Gw 2 (Remote domain: 172.44.266.44). Diversion

field has not the canonical form: 1481001

Diagnosis: Check NPD configuration, Diversion filed

should be as follow: +331481001(canonical format)

corresponds to +33 (France Country Code) 1481001

(Forwarded device number)

Solution: Configure a NPD for normal calls and a NPD for

forward as below:

Here is NPD for normal calls:

┌─Consult/Modify: Numbering Plan Description (NPD)────

──────┐

Node Number (reserved) : 1

Instance (reserved) : 1

Instance (reserved) : 1

Description identifier : 100

Name : SIP

Ed. 07

138

TG0069

OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

(https://www.facebook.com/sharer.php?src=sp&u=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2Fdoc%2F6856695%2Ftrou
│

Calling Numbering plan ident. + NPI/TON Isdn National

(https://twitter.com/intent/tweet?text=TROUBLESHOOTING%20GUIDE&url=https%3A%2F%2Ffanyv88.com%3A443%2Fhttps%2Fmanualzilla.com%2F
Called numbering plan ident. + NPI/TON : Isdn Unknown

│ Authorize personal calling num use + True

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Install. number source + NPD source

Default number source + None used

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│

Called DID identifier : 10

│ Calling/Connected DID identifier : -1

Installation number : 9839

└─────────────────────────────────

And this is NPD for fwd calls:

┌─Consult/Modify: Numbering Plan Description (NPD)────

──────┐

Node Number (reserved) : 1

Instance (reserved) : 1

Instance (reserved) : 1

Description identifier : 69

Name : FWD

Calling Numbering plan ident. + Unknown

Called numbering plan ident. + Unknown

│ Authorize personal calling num use + False

Install. number source + None used

Default number source + None used

Called DID identifier : 10

│ Calling/Connected DID identifier : 10

└────────────────────────────────────────┘

11.12.21

Leg1 and leg2 are external set, when OXE user performs a
blind transfer, it doesn’t

work

Symptom: External UserA calls OXE user B thru public SIP

Trunk(OXE user DDI: 210457060).

User B calls C (mobile phone) through public SIP trunk

B transfers the call to A before C answers

C answers the call but is not able to talk to external

user, transfer is not complete by OXE

Diagnosis: Parameter “Support Re-Invite without SDP” is

checked at TRUE on SIP External Gateway.

Consequence is OXE doesn‟t perform transfer due to a R&D

restriction on support of PRACK by remote

 according to this OXE configuration.

Solution: When PRACK is supported by SIP Provider, the

parameter “Support Re-Invite without SDP” must

be checked at false on SIP External Gateway.

11.12.22

SingleStep Transfer with REFER, no referred-by in the

following INVITE

Symptom: OXE user A makes a call to an external SIP

Server user B through SIP ABC-F Trunk. SIP Server

user B makes a single step transfer to SIP Server user C

with REFER method. In the following INVITE sent

by OXE, the header referred-by is missing (see RFC 3892)

Solution: Since 10.1 (J2.501.21 release), a new

parameter is available on System > Other System Param >

SIP Parameters > Transfer : Refer using single step.

This paramter is set by default at True and to obtain

Referred-by in such case, it must be checked at False.

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Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

11.13 Summary for SIP issue analyse

The purpose of this chapter is give a way to analyse a

SIP issue.

In case of SIP issue, a minimum of traces must be done,

the “motortrace” trace is the minimum to make. The

Infocollect must be done every time in case of SIP issue

to get all the information needed to troubleshoot.

Here the different steps to start the analyse:

Check if the SIP equipment is validated by Alcatel-

Lucent.

Check if the OXE configuration and SIP equipments

respect the rules given on this

document.

Check if the CPUs belong to the IP domain 0.

Check the “Network” management.

Check the local SIP configuration (motortrace c).

Check the incvisu file, and if SIP incidents, check the

sipalarm files to find the causes of

them.

Check if an incident or a backtrace is generated when

the issue is present.

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Check if the problem is from the SIPMOTOR or the Call
Handling

If a SR will be openened:

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-

Ed. 07

Provide a minimum of traces.

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Provide the call scenario (Caller, Called Party, IP
addresses, etc...), provide all the

information you can.

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Provide the Infocollect.

Provide your anlayse of the issue, it is mandatory for

you to make an anlyse before to open

a SR.

Provide a remote connection to the site (RMA, VPN,

etc...)

140

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OmniPCX Enterprise

Session Iniation Protcol (SIP)

TROUBLESHOOTING GUIDE No.0069

BEFORE CALLING ALCATEL-LUCENT’S SUPPORT CENTER

Before calling Alcatel‟s Business Partner Support Centre

(ABPSC), make sure that you have read

through:

The Release Notes which lists features available,

restrictions etc.

This chapter and completed the actions suggested for

your system‟s problem.

Additionally, do the following and document the results

so that the Alcatel Technical Support can

better assist you:

Have any information that you gathered while

troubleshooting the issue to this point available to

provide to the TAC engineer (such as traces).

[Have a network diagram ready in case of ABC-F

Networking problem].

[Have a data network diagram ready in case of VoIP

problems. Make sure that relevant information

is listed such as bandwidth of the links, equipments

like firewalls, etc.].

[Have a VoIP Audit report available in case of VoIP

problems].

Note

Dial-in access is also mandatory to help with effective

problem resolution.

Comments

Adapt the paragraph if specific or additional

information or actions are required depending on the

subject.

End of document

Legal notice:

Alcatel, Lucent, Alcatel-Lucent and the Alcatel-Lucent

logo are trademarks of Alcatel-Lucent.

All other trademarks are the property of their

respective owners.

The information presented is subject to change without

notice.

Alcatel-Lucent assumes no responsibility for

inaccuracies contained herein.

 Copyright © 2013 Alcatel-Lucent. All rights reserved

Ed. 07

141

TG0069

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