SIPTRUNK.COM Configuration Guide for Cisco CallManager (Unified Communications Manager)

The following Cisco configuration sheet will enable you do the following:

1. Register Cisco user agent with the Siptrunk.com trunking service

2. Place an outbound call and be authenticated

3. Receive inbound calls

Note: This config is not a complete solution, it’s just the key parts for the above.

KEY POINTS:

1) You must modify the INVITE message to re-write the SIP header to use username@gw1.siptrunk.com or username@gw2.siptrunk.com (see below config) in order to use digest authentication.

2) Siptrunk.com trunking releases the media to the nearest carrier media gateway to you for optimal performance.  Therefore, there is no way of knowing what IP address the RTP will be coming from.  Its best to allow all UDP for testing, then if absolutely necessary, look to lock down the UDP range where the media is coming from.  (this can be 1024-65535)

Sample Siptrunk.com Cisco IOS Sample Config:

(REPEAT FOR GW2.SIPTRUNK.COM BACKUP SERVER)

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

 !Allow SIP to the router from siptrunk.com

voice service voip

ip address trusted list

ipv4 63.247.69.226

ipv4 205.251.137.154

 

!Configure SIP profile to modify the INVITE message.

!1. Replace the "username" with actual username

!2. Replace the "ip-address" with the IP address which shouldn't be there

!This will then ensure that all INVITE headers contain the username@gw1.siptrunk.com which is the second field (the replace field)

 

voice class sip-profiles 1

request INVITE sip-header From modify "<sip:username@ip-address>" "<sip:username@gw1.siptrunk.com>"

 

!Create translation rule to replace source number / extension number with siptrunk.com username

 

voice translation-rule 5

rule 1 /^.*/ /username/

voice translation-profile SIPTRUNK.COM-Outgoing

translate calling 5

 

!Configure SIP user agent

 

sip-ua

credentials username username password your-password realm gw1.siptrunk.com

retry invite 2

retry register 10

timers connect 100

registrar 1 dns:gw1.siptrunk.com expires 360 refresh-ratio 20 auth-realm gw1.siptrunk.com

 

!Create dial-peer for outgoing calls

 

dial-peer voice 2 voip

description **Outgoing Calls to SIPTRUNK.COM SIP Trunk**

translation-profile outgoing SIPTRUNK.COM-Outgoing

destination-pattern 9.T

session protocol sipv2

session target dns:gw1.siptrunk.com

codec g711ulaw

voice-class sip dtmf-relay force rtp-nte

voice-class sip profiles 1

dtmf-relay rtp-nte

no vad

authentication username username password your-password realm gw1.siptrunk.com

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