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Ans: The Four Things That Are Need To Be Kept in Mind Are A. Framing B. Line-Coding C. Switch Type D. Clock Source

The document discusses troubleshooting PRI/ISDN circuits and phone registration. It provides steps to troubleshoot PRI/ISDN circuits such as checking the show isdn status command output and verifying layer 1 and layer 2 statuses. It also describes the phone boot process and differences between SCCP and SIP phone registration. Key steps in phone registration include obtaining DHCP and TFTP information, downloading configuration files, and registering with CallManager. The document answers additional questions about call manager groups, debug commands, and media resources.

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80 views7 pages

Ans: The Four Things That Are Need To Be Kept in Mind Are A. Framing B. Line-Coding C. Switch Type D. Clock Source

The document discusses troubleshooting PRI/ISDN circuits and phone registration. It provides steps to troubleshoot PRI/ISDN circuits such as checking the show isdn status command output and verifying layer 1 and layer 2 statuses. It also describes the phone boot process and differences between SCCP and SIP phone registration. Key steps in phone registration include obtaining DHCP and TFTP information, downloading configuration files, and registering with CallManager. The document answers additional questions about call manager groups, debug commands, and media resources.

Uploaded by

sumit rustagi
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as DOCX, PDF, TXT or read online on Scribd
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1.

What are the four things that are need to be kept in mind while configuring the
PRI/ ISDN circuits?
Ans: The four things that are need to be kept in mind are
a. Framing
b. Line-Coding
c. Switch Type
d. Clock Source

2. How will you troubleshoot the PRI/ISDN circuits?


Ans: We will issue the “show isdn status” command and will try to look at the output:
Router#show isdn status

Global ISDN Switchtype = primary-qsig


ISDN Serial0/1/1:23 interface
dsl 0, interface ISDN Switchtype = primary-qsig
**** Slave side configuration ****
Layer 1 Status:
ACTIVE
Layer 2 Status:
TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED
Layer 3 Status:
0 Active Layer 3 Call(s)
Active dsl 0 CCBs = 0
The Free Channel Mask: 0x00000000
Number of L2 Discards = 0, L2 Session ID = 0
Total Allocated ISDN CCBs = 0

In a working scenario, please find the status as under:

Layer 1 status should be Active, in case if the Layer 1 Status is Deactivated or shutdown then
we will try to bounce the controller

Layer 2 status should be MULTIPLE_FRAME_ESTABLISHED in case if the layer 2 status is


shown as “TEI_UNASSIGNED” or “AWAITING_ESTABLISHMENT” please verify the
configuration of the controller and make sure the configuration is done as per the service
provider after verifying the config if you still face the issue then please reach out to the
service provider.

3. List the required debug commands for troubleshooting the ISDN signalling ?
Ans: Debug ISDN q931 is used to troubleshoot the layer 3 signalling and below things are
populated in the output:
a. Calling Number
b. Called Number
c. Disconnect Cause
d. Which party has disconnected the call

4. Difference Between CAS and CCS Signalling?


Ans: CAS is called as Channel Associated Signalling, the signalling information is transmitted
using same bandwidth as voice.
CAS is sometimes called robbed-bit signaling because user bandwidth is robbed by the network
for signaling. Every 8th bit of Sixth frame contains the signalling information like
Calling party Number and Called Party number, etc.

CCS is called as common channel signalling in which a dedicated channel is used for the
signalling information.

For example if a T1 controller is configured as a CAS circuit then all the 24 Channels will be used
and 8th bit of 6th frame of each channel will have the signalling information.

But in Case of CCS the T1 controller will have 23 channels dedicated for the calls and one
channel is dedicated for the signalling information which is sometimes called as the Delta
channel (Signalling information) and the channels which carry the calls 23 channels are called as
the Bearer Channels.

5. Phone Boot Process ?


Ans: Concepts of Phone Registration :

Steps:

a. Phone gets power from the switch


b. Phone boots from its current firmware
c. CDP OR LLDP is bi-directionally transmitted to determine the voice vlan phone sends
a DHCP request and gets a response from the DHCP server (DORA)
d. Phone contacts TFTP server (Configured as option 150 or option 66 in the DHCP)
with request for <Sep_macaddess>.cnf.xml file but for versions above cucm 8.X that
there is a feature called as SBD in that case it will look for
<SEP_macaddress>.cnf.xml.sgn file (signed versions). In case if the phone is not
configured on UCM but auto registration is enabled in that case default.cnf.xml file is
served.

Note: Auto registration is not possible in a Secure Cluster (Mixed Mode)

e. Phone parses the XML config file for <load_informaiton>(firmware version)& process
node_name( CPN)
f. Phone requests the new load file from the TFTP server in case the there is a
difference in the version)
g. Phone registers with the CUCM sever based on the DP (CUCM_ group)
h. Sever hands down all the config to the phone (ex: line (DN), Speed dials, Softkey
template etc.)

Graphical View:
Difference Between the SIP phone and SCCP phone registration:

A. SCCP Phone Registration Process

1. SCCP phone obtains the Power (PoE or AC adapter).


2. The phone loads its locally stored firmware image.
3. The phone learns the Voice VLAN ID via CDP from the switch.
4. The phone uses DHCP to learn its IP address, subnet mask, default gateway and TFTP
server address.
5. The phone contacts the TFTP server and requests its configuration file. Each phone has
a customized configuration file named SEP<mac_address>.cnf.xml created by CUCM
and uploaded to TFTP when the administrator creates or modifies the phone.
6. The phone registers with the primary CUCM server listed in its configuration file. CUCM
then sends the softkey template to the phone using SCCP messages.
What is in that SEP<mac_address>.cnf.xml file ?

This file contains a list of CUCM server, in order, that the phone should register with. It lists
teh TCP ports it should use for SCCP communication. It also lists the firmware version for
each device model and the service URLs that each device should be using.
The CUCM server sends other configurations such as DNs, softkeys and speed dials via the
SCCP messages in the last phase of the registration process.

B. SIP Phone Registration Process

SIP Phones use a different set of steps to achieve the same goal. Steps 1 to 4 are the same
as SCCP Phones

1. The phone contacts the TFTP server and requests the Certificate Trust List file (only if the
cluster is secured).
2. The phone contacts the TFTP server and requests its SEP<mac-address>.cnf.xml
configuration file.
3. If the SIP Phone has not been provisioned before boot time, the SIP Phone downloads the
default configuration XMLDefault.cnf.xml file from the TFTP server.
4. The SIP phone requests a firmware upgrade (Load ID file), if one was specified in the
configuration file. This process allows the phone to upgrade the firmware image
automatically when required for a new version of CUCM.
5. The phone downloads the SIP dial rules configured for that phone.
6. The phone Establish connection with the primary CUCM and the TFTP server end to end.
7. The phone Registers with the primary CUCM server listed in its configuration file.
8. The phone downloads the appropriate localization files from TFTP
9. The phone downloads the softkey configurations from TFTP
10. The phone downloads custom ringtones (if any) from TFTP.

In order to review the config file:

On the browser type: https://<ipaddressofthecucm>:6970/configfilecachelist.txt

Q6: How many servers can be defined in a call manager group and how the phone will try
to register to them in case if it loses connectivity to the primary CUCM ?

Ans: In a CUCM group a maximum of 3 CUCM servers can be defined at one time and then
the CUCM group in assigned to the Device pool and the phone gets to know the list of the
CUCM servers from the device pool.
The phone maintains a contact with the primary server and sends a keep alives to the CUCM
in case if it doesn’t have any reply in 30 seconds, then the phone moves to the another
CUCM server and tries to register it if in case it doesn’t have any reply in the next 30
seconds then it will move to the last server in the CUCM group if the connectivity to all the
CUCM server fails then it will try to register under SRST provided that the SRST Reference is
defined in the Device pool.

Q7: What are the difference between SIP and SCCP phone registration ?
Ans: The phone downloads the SIP dial rules configured for that phone.
The phone Establish connection with the primary CUCM and the TFTP server end to end.
The phone Registers with the primary CUCM server listed in its configuration file.
The phone downloads the appropriate localization files from TFTP
The phone downloads the softkey configurations from TFTP
The phone downloads custom ringtones (if any) from TFTP.

Q 8 : What kind of traces are needed while troubleshooting for the phone registration
issues?

Ans: Event Viewer System Logs


Event Viewer Application Logs:
Both of the above tells about the reason code why the phone/device went
unregistered
Cisco Call Manager traces : Will tell about the phone registration issues in detail
Console logs from the phone
Status messages: will give a glimpse why the registration failed.

Q9 : What is the difference between PIN and Password on the end user page ?
Ans: The Pin is the extension mobility pin that is issues to the User, where in the password is
used to login to the user page of the CUCM.

Q10 related to Q11: Can you change a password for the LDAP user ? Ans: NO

Q 12: What are Media resources, when are they used.

Ans: a. It can be a hardware or a software based entity


b,. It is used to merge multiple audio streams to create a single output stream

Types of media resources:


a. MTP: can be hardware or software based
1. DTMF conversion
2. Sip trunks in order to support early offer / on H323 Gw to support fast start
3. Used to convert G711u law to G711 A law
b. Conf Bridge :
1. Can be hardware of software (for software only supports G711 codec)
2. It is used for conferencing of calls
c. Transcoder : Only hardware, used for changing the form of codec from one form to
another. For ex: changing the G711 media stream to G 729 media stream
d. ANN : Software based, used for announcements.

Q13 : Difference between 180 and 183 in SIP ?


Ans : 180 ringing is received when a ringback tone needs to be played by the local resources,
where in 183 session progress is a message which is initiated when an announcement is
being played to the user by the called party.

For ex: when a call is made to the back instead of the ring back tone we hear IVR message
this scenario is also called as early media or media cut through.

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