CH A P T E R 11

Configuring SIP and TEL URL Support

The Configuring SIP and TEL URL Support feature enables Cisco gateways to direct incoming calls to
a voice application based on the Uniform Resource Locator (URL) and Tcl Interactive Voice Response
(IVR) 2.0 and VoiceXML applications to place outbound calls to a SIP or TEL URL. This feature
expands call-control capabilities by allowing voice applications to use URL destinations and by
implementing dialing plans using SIP or TEL URLs, rather than telephone numbers.

Note In this document, the terms uniform resource identifier (URI) and URL are used interchangeably.

Finding Feature Information

Your software release may not support all the features documented in this module. For the latest feature
information and caveats, see the release notes for your platform and software release. To find information
about the features documented in this module, and to see a list of the releases in which each feature is
supported, see the “Feature Information for Configuring SIP and TEL URL Support” section on page 31.
Use Cisco Feature Navigator to find information about platform support and Cisco software image
support. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn. An account on
Cisco.com is not required.

Contents
• Prerequisites for Configuring SIP and TEL URL Support, page 2
• Restrictions for Configuring SIP and TEL URL Support, page 2
• Information About SIP and TEL URL Support, page 2
• How to Configure SIP and TEL URL Support, page 3
• Configuration Examples for SIP and TEL URL Support, page 21
• Where to Go Next, page 30
• Additional References, page 31
• Feature Information for Configuring SIP and TEL URL Support, page 31

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Prerequisites for Configuring SIP and TEL URL Support

Prerequisites for Configuring SIP and TEL URL Support

• Install Cisco IOS Release 12.3(4)T or a later release on the voice gateway.
• Configure basic voice application functionality on the gateway. For information, see “Configuring
Basic Functionality for Tcl and VoiceXML Applications”.
• Write a Tcl IVR 2.0 script or VoiceXML document that implements Session Iniation Protocol (SIP)
or telephone (TEL) URL support. For information, see the Tcl IVR API Version 2.0 Programmer’s
Guide or Cisco VoiceXML Programmer’s Guide.

Restrictions for Configuring SIP and TEL URL Support

• Access is provided only to headers in the SIP Invite, Subscribe, and Notify messages and H.323
Setup message.
• SIP and TEL URLs are not supported destinations for H.450 or SIP call transfers or call forwarding
using Tcl IVR or VoiceXML applications.
• A destination URL can have a maximum length of 1024 bytes.
• H.323 rejects a call if the URL is more than 512 characters because H.225 limits the URL length to
512 bytes.

Information About SIP and TEL URL Support

To configure voice application enhancements on a Cisco gateway, you should understand the following
concepts:
• SIP and TEL URL Support for Voice Applications, page 2
• Benefits of SIP and TEL URL Support, page 3

SIP and TEL URL Support for Voice Applications

Cisco IOS Release 12.3(4)T allows Tcl IVR 2.0 and VoiceXML applications to place calls to SIP and
TEL URL destinations in addition to E.164 telephone numbers. For outgoing calls, the voice application
provides the destination as either an URL or an E.164 number. Incoming calls are matched to a dial peer
based on the URL, triggering the voice application that is configured in the dial peer. New dial-peer
matching rules are implemented for URLs. The standard dial-peer matching rules apply to telephone
number destinations. VoiceXML applications can use the transfer tag to place calls to URL destinations.
Tcl IVR 2.0 scripts can do a leg setup to a URL instead of a telephone number.
For configuration information, use the following references:
• To create a voice class that defines the criteria for matching a URL, see the “Configuring a Voice
Class URI” section on page 3.
• To assign the URL voice class to an inbound or outbound dial peer, see the “Configuring an Inbound
Dial Peer to Match on a URI” section on page 7 or the “Configuring an Outbound Dial Peer for URI
Destinations” section on page 15.
• For a description of the new dial-peer matching rules for URLs, see to the destination uri command
and incoming uri command in the Cisco IOS Voice Command Reference.

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• To enable voice applications to read and pass SIP headers, see the Cisco IOS SIP Configuration
Guide.
• To specify SIP and TEL URLs in a Tcl IVR 2.0 script or VoiceXML document, see the Tcl IVR API
Version 2.0 Programmer’s Guide or Cisco VoiceXML Programmer’s Guide.

Benefits of SIP and TEL URL Support

• Enhances call-control features by supporting SIP and TEL URL destinations for a VoiceXML
transfer or Tcl leg setup. Incoming calls can be directed to voice applications based on a SIP or TEL
URL, and outbound calls can be placed using a URL to route the call.
• Complies with the W3C VoiceXML 2.0 Working Draft requirement for TEL URL support, enabling
additional information such as the account number to be provided when a call is transferred to or
from a VoiceXML application.
• Allows dialing plans to be implemented using SIP URLs instead of telephone numbers, simplifying
the management of SIP telephony networks.

How to Configure SIP and TEL URL Support

This section contains the following procedures:
• Configuring a Voice Class URI, page 3 (required)
• Configuring an Inbound Dial Peer to Match on a URI, page 7 (required)
• Verifying Dial-Peer Configuration for an Incoming URI Based Dial-Peer Match, page 12 (optional)
• Configuring an Outbound Dial Peer for URI Destinations, page 15 (required)
• Verifying Dial-Peer Configuration for an Outgoing URI, page 16 (optional)

Configuring a Voice Class URI

To configure a URI voice class and assign it to an inbound or outbound dial peer, choose one of the
following tasks, depending on whether a call has a SIP or TEL URI:
• Configuring a Voice Class SIP URI, page 3
• Configuring a Voice Class TEL URL, page 6
You define sets of related dial-peer attributes in voice classes. The Configuring SIP and TEL URL
Support feature introduces a new type of voice class for URI attributes. A URI voice class defines the
criteria for matching the URI in a call. You can configure a voice class to match either the entire URI or
specific fields within the URI, but, not both. You then assign the voice class to an inbound or outbound
dial peer. Calls are matched to the dial peer based on the URI.

Configuring a Voice Class SIP URI

This section describes how to configure a voice class to match on a SIP URI.

SUMMARY STEPS

1. enable

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2. configure terminal
3. voice class uri tag sip
4. pattern uri-pattern
5. host hostname-pattern or host {ipv4:ipv4-address | ipv6:ipv6-address | dns:domain-name |
hostname-pattern}
6. user-id username-pattern
7. phone context context-pattern
8. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice class uri tag sip Enter voice class configuration mode for a SIP URI.

Example:
Router(config)# voice class uri ab200 sip
Step 4 pattern uri-pattern (Optional) Specifies a regular expression pattern for
matching the entire TEL URI.
Example: Note When you configure the pattern command, you
Router(config-voice-uri-class)# pattern ^408 cannot configure the phone number or phone
context command, because the pattern command
matches the entire URI.
Step 5 host hostname-pattern (Optional) Specifies a regular expression pattern for
or matching the hostname field in the SIP URI.
host {ipv4:ipv4-address | ipv6:ipv6-address | • Only one instance of the command can be configured at
dns:domain-name}
any given time.
Specifies the host command by assigning an IPv4 address,
Example: IPv6 address, or Domain Name Server (DNS) name. You
Router(config-voice-uri-class)# host server1
can specify up to ten instances of this command.
or
Note You can use either the host hostname or host
Router(config-voice-uri-class)# host
ipv4:10.0.0.0 {ipv4:ipv4-address | ipv6:ipv6-address |
dns:domain-name} command.
Step 6 user-id username-pattern (Optional) Specifies a regular expression pattern for
matching the username field in the SIP URI.
Example:
Router(config-voice-uri-class)# user-id elmo
Step 7 phone context context-pattern (Optional) Specifies that only calls with a matching
phone-context field in the URI are selected.
Example: Note You must use the phone context command with
Router(config-voice-uri-class)# phone context either the host or user-id command or both. Using
408 this command alone does not result in any matches
on the voice class.
Step 8 end Exits voice class URI configuration mode and enters
privileged EXEC mode.
Example:
Router(config-voice-uri-class)# end

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What to Do Next

• To use the URI voice class to handle incoming calls, continue with the “Configuring an Inbound Dial
Peer to Match on a URI” section on page 7.
• To use the URI voice class to handle outbound calls, continue with the “Configuring an Outbound
Dial Peer for URI Destinations” section on page 15.

Configuring a Voice Class TEL URL

This task describes how to configure a voice class to match a TEL URI.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice class uri tag tel
4. pattern uri-pattern
5. phone number phone-number-pattern
6. phone context context-pattern
7. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice class uri tag tel Enters voice class URI configuration mode for a TEL URI.

Example:
Router(config)# voice class uri ab200 tel
Step 4 pattern uri-pattern (Optional) Specifies a regular expression pattern for
matching the entire TEL URI.
Example: Note If you use the pattern command you cannot use the
Router(config-voice-uri-class)# pattern ^408 phone number or phone context command in the
following steps because the pattern command
matches the entire URI.
Step 5 phone number phone-number-pattern (Optional) Specifies a regular expression pattern for
matching the phone number field in the TEL URI.
Example:
Router(config-voice-uri-class)# phone number
^8555
Step 6 phone context context-pattern (Optional) Specifies that only calls with a matching
phone-context field are selected.
Example: Note You must use the phone context command with the
Router(config-voice-uri-class)# phone context phone number command. Using this command
555 alone does not result in any matches on the voice
class.
Step 7 end Exits voice class configuration mode and enters privileged
EXEC mode.
Example:
Router(config-voice-uri-class)# end

Configuring an Inbound Dial Peer to Match on a URI

To configure an inbound dial peer to match an URI, choose one of the following tasks, depending on
whether the dial peer uses SIP or H.323.
• Configuring an Inbound Dial Peer to Match the URI in H.323 Calls, page 8
• Configuring an Inbound Dial-Peer to Match the URI on SIP Calls, page 9

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Prerequisites
• Enable SIP header passing. For information, see the Cisco IOS SIP Configuration Guide, Release
15.1
• Write a Tcl IVR 2.0 script or VoiceXML document that accepts a SIP or TEL URI. For information,
see the Tcl IVR API Version 2.0 Programmer’s Guide or Cisco VoiceXML Programmer’s Guide.

Configuring an Inbound Dial Peer to Match the URI in H.323 Calls

This task describes how to configure an inbound dial peer to match the URI in a H.323 call.

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag voip
4. service name
5. incoming uri {called | calling} tag
6. end

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DETAILED STEPS.

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial-peer configuration mode for a VoIP dial peer.

Example:
Router(config)# dial-peer voice 2 voip
Step 4 service name Associates an application with this VoIP dial peer.
Note When you configure the pattern command, you
Example: cannot configure the phone number or phone
Router(config-dial-peer)# service vapp1 context command, because the pattern command
matches the entire URI.
Step 5 incoming uri {called | calling} tag Specifies the URI voice class that matches calls to this dial
peer based on the destination URI or source URI in the
H.225 message.
Example:
Router(config-dial-peer)# incoming uri called Note This URI voice class must already be configured by
ab400 using the voice class uri command.

Note Incoming calls with a URI that matches the

configured voice class are linked to the dial peer and
to the VoiceXML or Tcl IVR application that you
configured in Step 4.
Step 6 end Exits the dial peer configuration mode and enters privileged
EXEC mode.
Example:
Router(config-dial-peer)# end

Configuring an Inbound Dial-Peer to Match the URI on SIP Calls

This task describes how to configure an inbound dial-peer to match the URI in a SIP call.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice class uri tag sip
4. host hostname-pattern or host {ipv4:ipv4-address | ipv6:ipv6-address | dns:dns-address |
hostname-pattern}
5. exit

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6. dial-peer voice tag voip

7. session protocol sipv2
8. incoming uri {from | request | to | via} tag
9. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice class uri tag sip‘ Creates a voice class for matching dial peers to a SIP and
enters voice URI class configuration mode.
Example:
Router(config)# voice class uri ab200 sip
Step 4 host hostname-pattern (Optional) Specifies a regular expression pattern for
or matching the hostname field in the SIP URI.
host {ipv4:ipv4-address | ipv6:ipv6-address | • This command can have a single instance only.
dns:domain-name}
Specifies the host command by assigning the IPv4 address,
IPv6 address, or DNS name.
Example:
Router(config-voice-uri-class)# host server1 • You can specify up to ten instances of this command.
or Note You can use either the host hostname or host
Router(config-voice-uri-class)# host {ipv4:ipv4-address | ipv6:ipv6-address |
ipv4:10.0.0.0 dns:domain-name} command.
Step 5 exit Enters global configuration mode.

Example:
Router(config-voice-uri-class)# exit
Step 6 dial-peer voice tag voip Enters dial peer voice configuration mode.

Example:
Router(config)# dial-peer voice 6000 voip
Step 7 session protocol sipv2 Configures SIP as the session protocol type.

Example:
Router(config-dial-peer)# session protocol
sipv2
Step 8 incoming uri {from | request | to | via} tag Specifies the voice class used to match a VoIP dial peer to
the URI of an incoming call.
Example:
Router(config-dial-peer)# incoming uri via
ab200
Step 9 end Exits dial peer voice configuration mode and enters
privileged EXEC mode.
Example:
Router(config-dial-peer)# end

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Verifying Dial-Peer Configuration for an Incoming URI Based Dial-Peer Match

Perform this task to display information about the dial-peer configuration for an incoming URI based
dial-peer match. These show commands need not be entered in any specific order.

SUMMARY STEPS

1. show voice class uri [tag | summary]

2. show dial-peer voice
3. show dialplan incall uri sip {from | request | to | via} uri

DETAILED STEPS

Step 1 show voice class uri [tag | summary]

Use the show voice class uri summary command to display summary information about the configured
URI voice classes. The following sample output shows the configured host instances under the URI voice
class:
Router# show voice class uri 1
Voice URI class: 1
SNMP status = Active
Schema = sip
user-id =
host =
phone context =
Host instances:
ipv4:10.0.0.0
ipv6:[2001:0DB8:0:1:FFFF:1234::5]
dns:ogw.example.com
exam.example.com

Step 2 show dial-peer voice

Use the show dial-peer voice command to display information for voice dial peers. The following
sample output shows the voice class URI 2 being set into dial peer 2:
Router# show dial-peer voice 2
VoiceOverIpPeer2
description = `',
tag = 2, destination-pattern = `',
URI classes:
Incoming (Request) =
Incoming (Via) = 1
Incoming (To) =
Incoming (From) =
Destination

Step 3 show dialplan incall uri sip {from | request | to | via} uri
Use the show dialplan incall uri to display the dial peer that is matched for a specific URI in an
incoming voice call. The following sample output shows the dial peer that is matched for an incoming
call, based on the selected URI:
Router# show dialplan incall uri sip via sip:9.13.38.83
Inbound VoIP dialpeer matching based on SIP URI's

VoiceOverIpPeer100
peer type = voice, system default peer = FALSE, information type = voice,
description = `',

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tag = 100, destination-pattern = `',

voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
rtp-ssrc mux = system
source carrier-id = `',target carrier-id = `',
source trunk-group-label = `',target trunk-group-label = `',
numbering Type = `unknown'
group = 100, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
modem transport = system,
URI classes:
Incoming (Request) =
Incoming (Via) = 1
Incoming (To) =
Incoming (From) =
Destination =
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
outgoing LPCOR:
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
mailbox selection policy: none
type = voip, session-target = `',
technology prefix:
settle-call = disabled
ip media DSCP = ef, ip media rsvp-pass DSCP = ef
ip media rsvp-fail DSCP = ef, ip signaling DSCP = af31,
ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41
ip video rsvp-fail DSCP = af41,
ip defending Priority = 0, ip preemption priority = 0
ip policy locator voice:
ip policy locator video:
UDP checksum = disabled,
session-protocol = sipv2, session-transport = system,
req-qos = best-effort, acc-qos = best-effort,
req-qos video = best-effort, acc-qos video = best-effort,
req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0,
req-qos video def bandwidth = 384, req-qos video max bandwidth = 0,
RTP dynamic payload type values: NTE = 101
Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,
A-law=8, GSMAMR-NB=117 iLBC=116, AAC-ld=114, iSAC=124
lmr_tone=0, nte_tone=0
h263+=118, h264=119
G726r16 using static payload
G726r24 using static payload
RTP comfort noise payload type = 19
fax rate = voice, payload size = 20 bytes
fax protocol = system

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fax-relay ecm enable

Fax Relay ans enabled
Fax Relay SG3-to-G3 Enabled (by system configuration)
fax NSF = 0xAD0051 (default)
codec = g729r8, payload size = 20 bytes,
video codec = None
voice class codec = `'
voice class sip session refresh system
voice class sip rsvp-fail-policy voice post-alert mandatory keep-alive interval
30
voice class sip rsvp-fail-policy voice post-alert optional keep-alive interval 30
voice class sip rsvp-fail-policy video post-alert mandatory keep-alive interval
30
voice class sip rsvp-fail-policy video post-alert optional keep-alive interval 30
text relay = disabled
Media Setting = forking (disabled) flow-through (global)
Expect factor = 10, Icpif = 20,
Playout Mode is set to adaptive,
Initial 60 ms, Max 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Fax nominal 300 ms
Max Redirects = 1, signaling-type = cas,
VAD = enabled, Poor QOV Trap = disabled,
Source Interface = NONE
voice class sip url = system,
voice class sip tel-config = system,
voice class sip rel1xx = system,
tvoice class sip outbound-proxy = system,
voice class sip asserted-id = system,
voice class sip privacy = system,
voice class sip e911 = system,
voice class sip history-info = system,
voice class sip pass-thru headers = system,
voice class sip pass-thru content unsupp = system,
voice class sip pass-thru content sdp = system,
voice class sip anat = system,
voice class sip g729 annexb-all = system,
voice class sip early-offer forced = system,
voice class sip negotiate cisco = system,
voice class sip reset timer expires 183 = system,
voice class sip block 180 = system,
voice class sip block 181 = system,
voice class sip block 183 = system,
voice class sip preloaded-route = system,
voice class sip random-contact = system,
voice class sip random-request-uri validate = system,
voice class sip call-route p-called-party-id = system,
voice class sip call-route history-info = system,
voice class sip privacy-policy send-always = system,
voice class sip privacy-policy passthru = system,
voice class sip privacy-policy strip history-info = system,
voice class sip privacy-policy strip diversion = system,
voice class sip bandwidth audio = system,
voice class sip bandwidth video = system,
voice class sip error-code-override options-keepalive failure = system,
voice class sip encap clear-channel = system,
voice class sip map resp-code 181 = system,
voice class sip bind control = system,
voice class sip bind media = system,
voice class sip authenticate redirecting-number = system,
redirect ip2ip = disabled
local peer = false
probe disabled,
Secure RTP: system (use the global setting)

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voice class perm tag = `'

Time elapsed since last clearing of voice call statistics never
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.
Last Disconnect Time = 0.
Matched: Digits: 0
Target:

Configuring an Outbound Dial Peer for URI Destinations

This task describes how to configure an outbound dial peer for calls placed to an URI destination.

Prerequisites
Write a Tcl IVR 2.0 script or VoiceXML document that implements call transfer to a SIP or TEL URI.
For information, see the TCL IVR API Version 2.0 Programming Guide or Cisco VoiceXML
Programmer’s Guide.

Restrictions
• Outbound calls to a SIP URL cannot be placed to a public switched telephone network (PSTN) leg
because dial-peer matching does not support this. Outbound calls to a TEL URL can be placed to a
PSTN call leg. The telephone number is pulled from the URL and used as the destination number.
• Underscores are not supported in the From header or in the callinfo originationNum field of a SIP
URI in a Tcl script or VoiceXML document.

SUMMARY STEPS

1. enable
2. configure terminal
3. dial-peer voice tag voip
4. session target ipv4:ip-address
5. session protocol sipv2
6. destination uri tag
7. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 dial-peer voice tag voip Enters dial peer configuration mode for a VoIP dial peer.

Example:
Router(config)# dial-peer voice 2 voip
Step 4 session target ipv4:ip-address (Optional) Specifies the IP address of the terminating
gateway
Example: Note The session target command is required when
Router(config-dial-peer)# session target placing a call to a TEL URI; it is optional for a SIP
ipv4:10.10.1.1 URI. If you do not configure a session target for a
SIP URI, the call is sent to the host address in the
SIP URI that is passed from the application
initiating the outbound call.
Step 5 session protocol sipv2 (Optional) Specifies the session protocol if you are using
SIP for calls between the local and remote gateways.
Example: Note If you are using H.323 do not configure the session
Router(config-dial-peer)# session protocol protocol command.
sipv2
Step 6 destination uri tag Specifies the URI class that links voice calls to this dial
peer.
Example: Note Before configuring the destination uri command,
Router(config-dial-peer)# destination uri 700 you should configure the URI voice class by using
the voice class uri command.
Step 7 end Exits dial peer voice configuration mode and enters
privileged EXEC mode.
Example:
Router(config-dial-peer)# end

Verifying Dial-Peer Configuration for an Outgoing URI

Perform this task to display information about the dial-peer configuration for an outgoing URI. These
show commands need not be entered in any specific order.

SUMMARY STEPS

1. show running-config

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2. show voice class uri [tag | summary]

3. show dialplan uri h323 uri

DETAILED STEPS

Step 1 show running-config

Use the show running-config command to display the configuration of the URI voice class and the
outbound dial peer.
The following example shows that voice class 500 is assigned to dial peer 599, which matches the calls
that have a pattern beginning with 123 in the outgoing SIP URI:
Router# show running-config
.
.
.
voice class uri 500 sip
pattern 123...
!
!
dial-peer voice 599 voip
session protocol sipv2
session target ipv4:10.10.1.1
destination uri 500
codec g711ulaw
!
Step 2 show voice class uri [tag | summary]
Use the show voice class uri command to display the configuration of the voice class:
Router# show voice class uri 500

Voice URI class: 500

Schema = sip
pattern = 123...

To display a list of all voice classes, use the show voice class uri summary command.
Router# show voice class uri summary
__________________________________________
Class Name Schema
------------------------------------------
100 sip
300 tel
500 sip
__________________________________________

Step 3 show dialplan uri

Use the show dialplan uri command to verify that the correct dial peer is matched for the URI.
The following example shows that dial peer 599 is linked to voice class 500, which matches the specified
URI. It also shows that the operational state of the dial peer is up:
Router# show dialplan uri sip:123456

Outbound dialpeer matching based on destination URI

VoiceOverIpPeer599
peer type = voice, information type = voice,
description = `',
tag = 599, destination-pattern = `',

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How to Configure SIP and TEL URL Support

answer-address = `', preference=0,

CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 599, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
DTMF Relay = disabled,
modem transport = system,
URI classes:
Incoming (Request) =
Incoming (To) =
Incoming (From) =
Destination = 500
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
Translation profile (Incoming):
Translation profile (Outgoing):
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
type = voip, session-target = `10.10.1.1',
technology prefix:
settle-call = disabled
ip media DSCP = ef, ip signaling DSCP = af31, UDP checksum = disabled,
session-protocol = sipv2, session-transport = system, req-qos = best-ef
acc-qos = best-effort,
RTP dynamic payload type values: NTE = 101
Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
CAS=123, ClearChan=125, PCM switch over u-law=0,A-law=8
RTP comfort noise payload type = 19
fax rate = voice, payload size = 20 bytes
fax protocol = system
fax-relay ecm enable
fax NSF = 0xAD0051 (default)
codec = g711ulaw, payload size = 20 bytes,
Expect factor = 0, Icpif = 20,
Playout Mode is set to default,
Initial 60 ms, Max 300 ms
Playout-delay Minimum mode is set to default, value 40 ms
Fax nominal 300 ms
Max Redirects = 1, signaling-type = ext-signal,
VAD = enabled, Poor QOV Trap = disabled,
Source Interface = NONE
voice class sip url = system,
voice class sip rel1xx = system,
voice class perm tag = `'
Time elapsed since last clearing of voice call statistics never
Connect Time = 0, Charged Units = 0,
Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0
Accepted Calls = 0, Refused Calls = 0,
Last Disconnect Cause is "",
Last Disconnect Text is "",
Last Setup Time = 0.
Matched: Digits: 0
Target:

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How to Configure SIP and TEL URL Support

Note For a description of the fields shown in this output, see the show dialplan uri command in the
Cisco IOS Voice Command Reference.

Troubleshooting Tips
• Do not use underscores in the From header or in the callinfo originationNum field of a SIP URI in
a Tcl script or VoiceXML document. Using underscores in these fields, as shown in the following
examples, causes call transfer to fail:
set callInfo(originationNum) “sip:[email protected]”

set headers(From) “sip:firstname_lastname”

• Use Tcl puts commands or VoiceXML log commands in your script to help with debugging. To
display the output from these commands, use the debug voip ivr script command for Tcl IVR 2.0
scripts, or the debug vxml puts command for VoiceXML documents.
For information on using the Tcl puts command, see the TCL IVR API Version 2.0 Programming
Guide. For information about the VoiceXML log command, see the Cisco VoiceXML Programmer’s
Guide.

Troubleshooting URI Matching

Perform this task to troubleshoot URI matching. These debug commands need not be entered in any
specific order.

SUMMARY STEPS

1. debug dialpeer
2. debug voice uri
3. debug voice ccapi error

DETAILED STEPS

Step 1 debug dialpeer

Use the debug dialpeer command to determine whether there is a dial peer with a voice class that
matches the URI in the Tcl script or VoiceXML document.
In the following example, the incoming call is matched to dial peer 1000, then the gateway searches
through the dial peers looking for a voice class that matches the URI in the outbound script:
Router# debug dialpeer

dialpeer detailed info debugging is on

*Mar 1 00:56:40.711: dpAssociateIncomingPeerSPI:

*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match rule (DP_MATCH_INCOMING_DNIS)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match incoming called number; called
(50267)
*Mar 1 00:56:40.711: dpMatchCore:
*Mar 1 00:56:40.711: dpMatchCore: dialstring(50267); expanded string(50267); calling()

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How to Configure SIP and TEL URL Support

*Mar 1 00:56:40.711: FillTarget: pDest(50267T) pPatn(50267)

*Mar 1 00:56:40.711: MatchNextPeer: peer 1000 matched
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Result=0 after DP_MATCH_INCOMING_DNIS;
peers (0x62F2E454)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerSPI:
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match rule (DP_MATCH_INCOMING_DNIS)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match incoming called number; called
(50267)
*Mar 1 00:56:40.711: dpMatchCore:
*Mar 1 00:56:40.711: dpMatchCore: dialstring(50267); expanded string(50267); calling()
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Result=-1 after DP_MATCH_INCOMING_DNIS;
peers (0x0)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match rule (DP_MATCH_ANSWER)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match answer address; calling (50006)
*Mar 1 00:56:40.711: dpMatchCore:
*Mar 1 00:56:40.711: dpMatchCore: dialstring(); expanded string(); calling(50006T)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Result=-1 after DP_MATCH_ANSWER; peers
(0x0)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match rule (DP_MATCH_ORIGINATE)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match destination pattern; calling
(50006)
*Mar 1 00:56:40.711: dpMatchCore:
*Mar 1 00:56:40.711: dpMatchCore: dialstring(); expanded string(); calling(50006T)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Result=-1 after DP_MATCH_ORIGINATE;
peers (0x0)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match rule (DP_MATCH_PORT)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Result=-1 after DP_MATCH_PORT; peers
(0x0)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match rule (DP_MATCH_SRC_CARRIER)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Result=-1 after DP_MATCH_SRC_CARRIER;
peers (0x0)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerSPI:
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match rule (DP_MATCH_INCOMING_DNIS)
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Match incoming called number; called
(50267)
*Mar 1 00:56:40.711: dpMatchCore:
*Mar 1 00:56:40.711: dpMatchCore: dialstring(50267); expanded string(50267); calling()
*Mar 1 00:56:40.711: FillTarget: pDest(50267T) pPatn(50267)
*Mar 1 00:56:40.711: MatchNextPeer: peer 1000 matched
*Mar 1 00:56:40.711: dpAssociateIncomingPeerCore: Result=0 after DP_MATCH_INCOMING_DNIS;
peers (0x62F2E454)
*Mar 1 00:56:49.203: dpMatchPeersMoreArg:
*Mar 1 00:56:49.203: dpMatchPeersCore:
*Mar 1 00:56:49.203: dpMatchPeersCore: Match rule (DP_MATCH_DEST_URI_AND_TGT_CARRIER)
*Mar 1 00:56:49.203: dpMatchPeersCore: Result=-1 after DP_MATCH_DEST_URI_AND_TGT_CARRIER
*Mar 1 00:56:49.203: dpMatchPeersCore: Match rule (DP_MATCH_DEST_AND_TGT_CARRIER)
*Mar 1 00:56:49.203: dpMatchPeersCore: Result=-1 after DP_MATCH_DEST_AND_TGT_CARRIER
*Mar 1 00:56:49.203: dpMatchPeersCore: Match rule (DP_MATCH_DEST_URI)
*Mar 1 00:56:49.203: dpMatchPeersCore: Match Dest. URI; URI
(sip:[email protected]?Subject=Hello&Priority=Urgent&testID=AL_FEAT_SIP_URL_O_RV_11
)
*Mar 1 00:56:49.203: dpMatchCore:
*Mar 1 00:56:49.203: dpMatchCore: dialstring(); expanded string(); calling()
*Mar 1 00:56:49.203: dpMatchPeersCore: Result=-1 after DP_MATCH_DEST_URI
*Mar 1 00:56:49.203: dpMatchPeersCore: Match rule (DP_MATCH_DEST)
*Mar 1 00:56:49.203: dpMatchPeersCore: Result=-1 after DP_MATCH_DEST
*Mar 1 00:56:49.203: dpMatchPeersCore: Match rule (DP_MATCH_TGT_CARRIER)
*Mar 1 00:56:49.203: dpMatchPeersCore: Result=-1 after DP_MATCH_TGT_CARRIER
*Mar 1 00:56:49.203: //-1//TCL2:HN0033E40C:/tcl_PutsCmd:
STATUS=ls_004
*Mar 1 00:56:49.203:

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Configuration Examples for SIP and TEL URL Support

A message showing Result=-1 indicates that a dial peer was not matched. Because this output does not
display another dial peer match statement, you know that the gateway failed to find a match.
Step 2 debug voice uri
Use the debug voice uri command to verify whether there is a voice class that matches the URI in the
Tcl script or VoiceXML document.
In the following example, the gateway failed to match the URI in the script to the only configured
voice class, 805:
Router# debug voice uri

Voice URI debugging is enabled

Router#
*Mar 1 01:09:03.319: vuri_match_class: tag (805)
*Mar 1 01:09:03.319: vuri_match_class_sip: Match with phone context
*Mar 1 01:09:03.319: vuri_match_class_sip: input ()
*Mar 1 01:09:03.319: vuri_match_class_sip: Match with host
*Mar 1 01:09:03.319: vuri_match_class_sip: input (example.com)
*Mar 1 01:09:03.319: vuri_match_class_sip: Match with user-id
*Mar 1 01:09:03.319: vuri_match_class_sip: input (xxx_no_under)
*Mar 1 01:09:03.319: vuri_match_class_sip: Match failed

Step 3 debug voice ccapi error

Use the debug voice ccapi error command to display any errors that might occur during the call transfer:
Router# debug voice ccapi error

*Mar 1 01:39:59.319: //32/7FAB684B8022/CCAPI/cc_get_associated_stream: Matched the

Stream CallID 0x21::
*Mar 1 01:40:04.279: //32/7FAB684B8022/CCAPI/cc_api_call_disconnect_done:
cause=28,retry=0,vcCauseCode=0

Any output from this command means that the call transfer did not complete, although the reason for the
error might not be obvious from the messages.

Configuration Examples for SIP and TEL URL Support

This section provides the examples that show the configuration for an originating and terminating
gateway, each configured to handle both SIP and TEL URIs.
• Example: Outbound Dial-Peer Originating Gateway, page 21
• Example: Outbound Dial-Peer Terminating Gateway, page 26

Note For information about reading and passing SIP headers, see the Cisco IOS SIP Configuration Guide.

Example: Outbound Dial-Peer Originating Gateway

This following sections provide examples of SIP and TEL URIs with Header passing using the respective
protocols in the originating gateway.
• Example: URIs with Header Passing Using SIP Protocol and H.323 Protocol, page 27

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Configuration Examples for SIP and TEL URL Support

Example: URIs with Header Passing Using SIP Protocol and H.323 Protocol
In the following example, when a SIP call comes into the originating gateway, the application
sip_headers_tcl asks the caller to enter an account number. After the account number is collected, it is
assigned to a header named AccountInfo. The AccountInfo is passed to the leg setup along with other
standard and user-defined headers in the destination URL.
The outgoing call matches on dial peer 766, which is configured with the destination uri command to
match on voice class 766. Voice class 766 is configured with the voice class uri command to match
destination SIP URI sip:[email protected]. The gateway places a call to
sip:[email protected].
When the call setup request is received by the gateway, the headers that are passed from the application
are included in the SIP INVITE message.
In the following example, when an H.323 call comes into the originating gateway, the application
tel_headers_vxml asks the caller to enter an account number. After the account number is collected, it is
assigned to a header named AccountInfo. The AccountInfo is passed to the leg setup along with other
user-defined headers in the destination URL.
The outgoing call matches on dial peer 767, which is configured with the destination uri command to
match on voice class 767. Voice class 767 is configured with the voice class uri command to match
telephone number 555-0100 and a phone context of 408. The voice class can also match against a URL
pattern. This example uses the phone number and phone context as a matching criteria. The gateway
places a call to a TEL URL with the header of
tel:555-0100;phone-context=408;tsp=example.com;Subject=HelloTelVXML;[email protected];
From=nobody; Priority=urgent'+';AccountInfo='+acctInfo.
When the call setup request is received by the gateway, the destination URL with headers and parameters
is passed in the setup message as part of the destination address.
!
version x.x
service timestamps debug datetime msec localtime
service timestamps log datetime msec localtime
no service password-encryption
service internal
!
hostname 10.7.102.32
!
no logging buffered
enable password lab
!
username 1111
username 2222 password 0 2222
!
!
resource-pool disable
!
aaa new-model
!
!
aaa authentication login h323 local group radius
aaa authorization exec h323 group radius
aaa accounting connection h323 start-stop group radius
aaa session-id common
ip subnet-zero
ip ftp username dump
ip ftp password dump123
ip host px1-sun 10.0.1.0
ip host rtsp-ws 10.1.1.0
ip host dev 10.3.0.1

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Configuration Examples for SIP and TEL URL Support

ip host px1-exm.example.com 10.0.1.1

ip host example.com 10.1.1.2
!
!
isdn switch-type primary-5ess
isdn voice-call-failure 0
!
!
voice service voip
sip
!
!
voice class uri 766 sip
pattern [email protected]*
!
voice class uri 767 tel
phone number 767....
phone context 408
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
ivr prompt memory 16384
ivr record memory system 256000
ivr record memory session 256000
ivr asr-server rtsp://nuance-asr/recognizer
ivr tts-server rtsp://nuance-asr/synthesizer
ivr record memory system 256000
ivr record memory session 256000
ivr asr-server rtsp://nuance-asr/recognizer
ivr tts-server rtsp://nuance-asr/synthesizer
rtsp client session history duration 1
rtsp client session history records 1
http client cache memory pool 0
http client cache memory file 10000
http client cache refresh 30
http client connection timeout 10
http client response timeout 10
fax interface-type modem
mta receive maximum-recipients 0
call-history-mib retain-timer 0
call-history-mib max-size 0
!
controller T1 0
framing esf
clock source line primary
linecode b8zs
cablelength short 133
pri-group timeslots 1-24
!
controller T1 1
framing esf
clock source line secondary 1
linecode b8zs
cablelength short 133
pri-group timeslots 1-24
!
controller T1 2
framing esf

Cisco IOS Tcl IVR and VoiceXML Application Guide

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Configuration Examples for SIP and TEL URL Support

clock source line secondary 2

linecode b8zs
cablelength short 133
pri-group timeslots 1-24
!
controller T1 3
framing esf
clock source line secondary 3
linecode b8zs
cablelength short 133
pri-group timeslots 1-24
!
gw-accounting h323
gw-accounting h323 vsa
gw-accounting voip
!
!
interface Ethernet0
ip address 209.165.201.1 255.255.255.224
ip helper-address 10.1.201.30
no ip route-cache
no ip mroute-cache
no cdp enable
!
interface Serial0
no ip address
shutdown
clockrate 2015232
no fair-queue
no cdp enable
!
interface Serial1
no ip address
shutdown
clockrate 2015232
no fair-queue
no cdp enable
!
interface Serial2
no ip address
shutdown
clockrate 2015232
no fair-queue
no cdp enable
!
interface Serial3
no ip address
shutdown
clockrate 2015232
no fair-queue
no cdp enable
!
interface Serial0:23
no ip address
no logging event link-status
dialer-group 1
isdn switch-type primary-5ess
isdn incoming-voice modem
isdn disconnect-cause 1
fair-queue 64 256 0
no cdp enable
!
interface Serial1:23
no ip address

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24 OL-11176-02
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Configuration Examples for SIP and TEL URL Support

no logging event link-status

isdn switch-type primary-5ess
no cdp enable
!
interface Serial2:23
no ip address
no logging event link-status
isdn switch-type primary-5ess
no cdp enable
!
interface Serial3:23
no ip address
no logging event link-status
isdn switch-type primary-5ess
no cdp enable
!
interface FastEthernet0
no ip address
shutdown
duplex auto
speed auto
no cdp enable
!
ip default-gateway 10.255.255.255
ip classless
ip route 209.165.200.20 255.255.255.224
no ip http server
ip pim bidir-enable
!
!
no cdp run
!
!
radius-server retransmit 3
radius-server authorization permit missing Service-Type
call rsvp-sync
!
application
service sip_headers_tcl tftp://dev/demo/TCL/scripts/sip_headers.tcl
paramspace english language en
paramspace english index 1
paramspace english location tftp://dev/demo/AUDIO/en/
!
service sip_headers_vxml tftp://dev/demo/VXML/scripts/sip_headers.vxml
paramspace english language en
paramspace english index 1
paramspace english location tftp://dev/demo/AUDIO/en/
!
service tel_headers_tcl tftp://dev/demo/TCL/scripts/tel_headers.tcl
paramspace english language en
paramspace english index 1
paramspace english location tftp://dev/demo/AUDIO/en/
!
service tel_headers_vxml tftp://dev/demo/VXML/scripts/tel_headers.vxml
paramspace english language en
paramspace english index 1
paramspace english location tftp://dev/demo/AUDIO/en/
!
!
voice-port 0:D
!
voice-port 1:D
!
voice-port 2:D

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 Chapter Configuring SIP and TEL URL Support
Configuration Examples for SIP and TEL URL Support

!
voice-port 3:D
!
mgcp modem passthrough voip mode ca
no mgcp timer receive-rtcp
!
mgcp profile default
!
dial-peer cor custom
!
!
dial-peer voice 1 pots
service sip_headers_tcl
incoming called-number 52948
port 0:D
!
dial-peer voice 2 pots
application tel_headers_vxml
incoming called-number 52950
port 0:D
!
dial-peer voice 767 voip
session target ipv4:10.0.0.1
destination uri 767
codec g711ulaw
!
dial-peer voice 766 voip
session protocol sipv2
session target ipv4:10.0.0.1
destination uri 766
codec g711ulaw
!
dial-peer voice 7671234 voip
service get_headers_vxml out-bound
destination-pattern .......
session protocol sipv2
session target ipv4:10.0.0.1
codec g711ulaw
!
sip-ua
sip-server ipv4:10.0.1.1
!
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
!
exception core-file special3
exception dump 10.7.100.1
end

Example: Outbound Dial-Peer Terminating Gateway

This section provides examples of SIP and TEL URIs with Header passing using the respective protocols
in the terminating gateway.
• Example: URIs with Header Passing Using SIP Protocol and H.323 Protocol, page 27

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Example: URIs with Header Passing Using SIP Protocol and H.323 Protocol
In the following example, when the call arrives at the terminating gateway and dial peer 766 is matched,
the gateway stores all headers received in the incoming INVITE message so these can be accessed by
the application.
The inbound dial peer can be configured to match the request-URI, or the “To” or “From” header in the
incoming INVITE message. This example uses the request-URI for matching. The incoming call
matches on dial peer 766, which is configured with the incoming uri request command to match on
voice class 766. Voice class 766 is configured to match the incoming SIP request-URI
sip:[email protected].
When the call is handed to the application configured in the inbound dial peer, get_headers_tcl, this Tcl
application can read any header that is part of the incoming INVITE message.
In the following example, when the call arrives at the terminating gateway and dial peer 767 is matched,
the gateway stores the incoming URI so it can be accessed by the application.
The inbound dial peer can be configured to match the entire TEL URL pattern, the E.164 number portion,
or the phone context of the TEL URL. This example uses the phone number and phone context for
matching. The incoming call matches on dial peer 767, which is configured with the incoming uri called
command to match on voice class 767. Voice class 767 is configured to match the incoming called URL
with the header of
tel:555-0100;phone-context=408;tsp=example.com;Subject=HelloTelVXML;[email protected];
From=nobody;Priority=urgent'+';AccountInfo='+acctInfo.
When the call is handed to the application configured in the inbound dial peer, get_headers_vxml, this
VoiceXML application can read any header which is part of the incoming called URI received in the
setup indication.
!
version x.x
service timestamps debug uptime
service timestamps log uptime
no service password-encryption
service internal
!
hostname as5300-09
!
enable secret 5 $1$KRsb$cFAQFOylLr9j5Fof.eLgx1
enable password lab
!
!
!
resource-pool disable
clock timezone PDT -8
clock calendar-valid
!
ip subnet-zero
no ip domain lookup
ip domain name fieldlabs.example.com
ip host dcl1server 10.7.108.2
ip host px1-sun 10.14.99.1
ip host dirt 192.168.254.254
ip host jurai 192.168.254.254
ip host dclserver 10.7.108.2
ip host dcl2server 10.7.112.2
ip host ts 10.7.100.1
!
!
isdn switch-type primary-5ess
isdn voice-call-failure 0

Cisco IOS Tcl IVR and VoiceXML Application Guide

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 Chapter Configuring SIP and TEL URL Support
Configuration Examples for SIP and TEL URL Support

!
!
voice service voip
sip
header-passing
!
!
voice class uri 766 sip
pattern [email protected]*
!
voice class uri 767 tel
phone number 767....
phone context 408
!
!
!
!
!
no voice hpi capture buffer
no voice hpi capture destination
!
!
ivr record memory system 100000
ivr record memory session 100000
ivr record memory system 100000
ivr record memory session 100000
fax interface-type modem
mta receive maximum-recipients 0
!
controller T1 0
framing esf
clock source line primary
linecode b8zs
pri-group timeslots 1-24
!
controller T1 1
framing sf
clock source line secondary 1
linecode ami
!
controller T1 2
framing sf
linecode ami
!
controller T1 3
framing sf
linecode ami
!
!
!
interface Ethernet0
ip address 209.165.200.225 255.255.255.224
ip helper-address 192.168.254.254
no ip route-cache
no ip mroute-cache
no cdp enable
!
interface Serial0:23
no ip address
dialer-group 1
isdn switch-type primary-5ess
isdn incoming-voice modem
fair-queue 64 256 0
no cdp enable

Cisco IOS Tcl IVR and VoiceXML Application Guide

28 OL-11176-02
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Configuration Examples for SIP and TEL URL Support

!
interface FastEthernet0
ip address 209.165.201.28 255.255.255.224
no ip route-cache
no ip mroute-cache
duplex half
speed 10
no cdp enable
!
ip default-gateway 10.7.0.1
ip classless
ip route 209.168.201.1 255.255.255.224 10.165.196.1
ip route 209.168.201.5 255.255.255.224 10.165.0.1
no ip http server
ip pim bidir-enable
!
!
no cdp run
!
!
call rsvp-sync
!
application
service get_headers_tcl tftp://dev/demo/TCL/scripts/get_headers.tcl
paramspace english language en
paramspace english index 1
paramspace english location tftp://dirt/cchiu/AUDIO/en/
!
service voice get_headers_vxml tftp://dev/demo/VXML/scripts/get_headers.vxml
paramspace english language en
paramspace english index 1
paramspace english location tftp://dirt/cchiu/AUDIO/en/
!
!
voice-port 0:D
!
mgcp ip qos dscp cs5 media
mgcp ip qos dscp cs3 signaling
!
mgcp profile default
!
dial-peer cor custom
!
!
!
dial-peer voice 1 pots
service test
incoming called-number 52950
port 0:D
!
dial-peer voice 767 voip
service get_headers_vxml
session target ipv4:10.0.0.1
incoming uri called 767
codec g711ulaw
!
dial-peer voice 766 voip
service get_headers_tcl
session protocol sipv2
session target ipv4:10.0.0.0
incoming uri request 766
codec g711ulaw
!
dial-peer voice 2 pots

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 Chapter Configuring SIP and TEL URL Support
Where to Go Next

destination-pattern 767....
port 0:D
prefix 9767
!
sip-ua
!
!
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
password lab
login
!
scheduler interval 1000
end

Where to Go Next
• To configure properties for audio files, see “Configuring Audio File Properties for Tcl and
VoiceXML Applications”.
• To configure voice recording using a VoiceXML application, see “Configuring VoiceXML Voice
Store and Forward”.
• To configure properties for speech recognition or speech synthesis, see “Configuring ASR and TTS
Properties”.
• To configure a VoiceXML fax detection application, see “Configuring Fax Detection for
VoiceXML”.
• To configure telephony call-redirect features for voice applications, see “Configuring Telephony
Call-Redirect Features”.
• To configure session interaction for a Tcl IVR 2.0 application, see “Configuring Tcl IVR 2.0 Session
Interaction”.
• To monitor and troubleshoot voice applications, see “Monitoring and Troubleshooting Voice
Applications”.

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 Chapter Configuring SIP and TEL URL Support
Additional References

Additional References

Related Documents
Related Topic Document Title
Cisco IOS commands Cisco IOS Master Commands List, All Releases
Cisco IOS Voice commands Cisco IOS Voice Command Reference
Overview of Cisco Tcl IVR and VoiceXML Cisco IOS Tcl IVR and VoiceXML Application Guide
Applications

MIBs
MIB MIBs Link
• CISCO-VOICE-DIAL-CONTROL-MIB To locate and download MIBs for selected platforms, Cisco software
releases, and feature sets, use Cisco MIB Locator found at the
• CISCO-VOICE-DNIS-MIB
following URL:
http://www.cisco.com/go/mibs

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/cisco/web/support/index.html
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies.
To receive security and technical information about
your products, you can subscribe to various services,
such as the Product Alert Tool (accessed from Field
Notices), the Cisco Technical Services Newsletter, and
Really Simple Syndication (RSS) Feeds.
Access to most tools on the Cisco Support website
requires a Cisco.com user ID and password.

Feature Information for Configuring SIP and TEL URL Support

Table 11-1 lists the features in this module and provides links to specific configuration information.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which software images support a specific software
release, feature set, or platform. To access Cisco Feature Navigator, go to http://www.cisco.com/go/cfn.
An account on Cisco.com is not required.

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 Chapter Configuring SIP and TEL URL Support
Feature Information for Configuring SIP and TEL URL Support

Note Table 11-1 lists only the software release that introduced support for a given feature in a given software
release train. Unless noted otherwise, subsequent releases of that software release train also support that
feature.

Table 11-1 Feature Information for Configuring SIP and TEL URL Support

Feature Name Releases Feature Information

SIP and TEL URL support 12.3(4)T SIP and TEL URL support enables Cisco gateways to direct
12.3(14)T incoming calls to a voice application based on the URL and
Tcl IVR 2.0 and VoiceXML applications to place outbound
calls to a Session Initiation Protocol (SIP) or telephone
(TEL) URL. It expands call-control capabilities by allowing
voice applications to use URL destinations and dialing
plans to be implemented using SIP or TEL URLs rather than
telephone numbers.
The following sections provide information about this
feature:
• Configuring an Inbound Dial Peer to Match on a URI,
page 7
• Configuring an Outbound Dial Peer for URI
Destinations, page 15
Support Inbound Dial-peer Match Based on 15.1(2)T The Support Inbound Dial-peer Match Based on Remote IP
Remote IP Address on SIP Trunks Address on SIP Trunks feature supports the expansion of
inbound dial-peer matching logic to include matching based
on the source IP address of inbound signaling on a SIP
trunk. This feature enables enforcement of specific
call-treatment, security, and routing policies on each SIP
trunk.
The following section provide information about this
feature:
• Configuring an Inbound Dial-Peer to Match the URI on
SIP Calls, page 9
The following commands were introduced or modified:
dial-peer voice, voice-class uri.

Cisco and the Cisco Logo are trademarks of Cisco Systems, Inc. and/or its affiliates in the U.S. and other countries. A listing of Cisco's trademarks
can be found at www.cisco.com/go/trademarks. Third party trademarks mentioned are the property of their respective owners. The use of the word
partner does not imply a partnership relationship between Cisco and any other company. (1005R)

Any Internet Protocol (IP) addresses and phone numbers used in this document are not intended to be actual addresses and phone numbers. Any
examples, command display output, network topology diagrams, and other figures included in the document are shown for illustrative purposes only.
Any use of actual IP addresses or phone numbers in illustrative content is unintentional and coincidental.

© 2007–2010 Cisco Systems, Inc. All rights reserved.

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