Using SIP trunk to dial out on Call manager Express

by Ras 1. March 2012 21:57

Here is a sample if SIP trunk for Cisco Call manager express with outgoing dial-peers for Australia. Using SIP trunk for outgoing call is a good way to save money for companies.


### Specify voip service configuration

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
sip
registrar server expires max 3600 min 3600
localhost dns:nsit.local
no update-callerid
!

## Specify a DNS server for being able to resolve the voip provider dns name


ip domain name domain.local
ip name-server 192.168.0.10
!

## Specify codecs


voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!

## Specify translation rules for incoming and outgoing calls


voice translation-rule 1
rule 1 /^0/ //
!
voice translation-rule 2
rule 1 /.*/ /1300783899/
!
voice translation-rule 3
rule 1 /.*/ /290/
!
voice translation-rule 5
rule 1 /^ABCD\(.*\)/ /\1/
!
!
voice translation-profile INCOMING_TRANSLATION_PROFILE
translate called 3
!
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate calling 2
translate called 1
!
voice translation-profile SIP_Passthrough
translate called 5
!

## Configure the sip agent and voip provider settings


sip-ua
credentials username 1300783899 password 7 06081C28585D1C091518001F realm voip.comcen.com.au
authentication username 1300783899 password 7 11070A0C03011E1C14253930
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:voip.comcen.com.au expires 3600
sip-server dns:voip.comcen.com.au
host-registrar
!

## Specify dial plans for incoming and outgoing on SIP


dial-peer voice 110 voip
permission term
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
translation-profile incoming INCOMING_TRANSLATION_PROFILE
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 111 voip
corlist outgoing call-local
description ** star code to SIP trunk (Generic SIP Trunk Provider) **
destination-pattern *..
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 300 voip
description **Australia*All**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 1
destination-pattern 0T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad

Tags:

Cisco | Voice/Video

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About the author

Ras is a network/Security professional working on multiple areas with multiple certificates like CCNP, CCIP, CCSP, CCSA, CCSE, LPI, PM, IPv6, ..

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