SIP Trunk

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Jul 29th, 2010
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I want to setup basic Sip trunk with Service provider on my CME. There would be no authentication required for the sip trunk configuration, as per the service provider.

In the configuration on CME, do I need to configure the sip-ua OR do i only conifgure the dial-peer with session target pointing to SIP server.

Please suggest.


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csshenoy80 Thu, 07/29/2010 - 09:38
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In the configuration, there is sip-ua configured with authentication option. In case, there is no authentication, is the sip-ua still required to be conifgured on the cme?

Jaime Valencia Thu, 07/29/2010 - 10:01
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If you do not need authentication just don't configure that line.

I assume your telco provided a registrar server config so that you need to configure.

Configuring SIP Register Support



If this helps, please rate

csshenoy80 Thu, 07/29/2010 - 11:06
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What exactly does the sip-ua configuration provide... it provide the e.164 number registration with the registrar server? Do i need it in my case, where I need to send the calls to and receive calls from the Service Provider.

Steven Holl Thu, 07/29/2010 - 11:18
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sip-ua configured a SIP User Agent.  It's essentially used if you need to have the router act as a User Agent instead of a SIP proxy.  So for an instance where you need registration.  We do sandwich some global SIP configurations in here instead of 'voice service voip/sip' from time-to-time, though.

For your scenario, sip-ua is likely not necessary.

Steven Holl Thu, 07/29/2010 - 10:53
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To put it in a simple answer: if your SP doesn't require registration, you likekly don't need to configure a sip-ua at all.  You'll probably be okay with just a SIP dial-peer pointing to the destination.

If in doubt, and something isn't working as intended, run 'debug ccsip mess' and see what's happening between CME and the provider.

lavanyaps Mon, 08/02/2010 - 02:46
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For the Sip trunk with the Service Provider on CME, you can directly configure the SIP Routing and there is no need of authentication using sip-ua.

Router CME #

voice service voip


  bind control source-interface FastEthernet0/1


voice class codec 1

codec preference 1 g729r8

codec preference 2 g729br8


dial-peer voice 25 voip

description sip turnk sp

destination-pattern 2978

session protocol sipv2

session target ipv4:

incoming called-number .T

voice-class codec 1

no vad


This configuration will help you to enable the SIP Trunk with the Service Provider IP.

Pl rate if this helps you.. thankz


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