How to configure a SIP TKUNK

Unanswered Question
Oct 13th, 2008

I have a CME 4.2 with liksys phone registered, I can make calls between them,I can make calls to PSTN by means of E1 link, but now I'm carrying out test using a SIP provider but I can't even communicate with SIP proxy. I have already set dial-peer, sip-ua parameters etc...I follow some examples I found in cisco but my tests failed.

I was provided an IP (172.20.8.8) in the same segment the proxy is, the CME has 10.10.10.1

Does any of you have an actual example I can follow?

I have this problem too.
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bwilmoth Fri, 10/17/2008 - 09:39

SIP trunks are similar to a phone line, except that SIP trunks use the IP network, not the PSTN. In addition, SIP trunks permit the convergence of voice and data onto common all-IP connections. In order to access the IP network using an SIP trunk, it is necessary that configurations be made on the service provider, as well as on the customer side. Customers need to set and configure CME, which is the PBX that will interpret the SIP signal adequately and pass traffic successfully. The service provider needs to configure an SIP Proxy Server. However, SIP trunks are more complicated to establish than regular PSTN trunks. The reason is that a customer faces challenges in handling different interpretation and implementations of SIP by equipment vendors, delivering security, managing quality of service (QoS), enabling Network Address Translation (NAT) and firewall traversal, and ensuring carrier-grade reliability and continuity of service.

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