Do SIP trunks have to terminate to the CUCM? Or can the terminate to a gateway on the CUCM? Everything I have read indicates it terminates to the CUCM directly, and I have a couple working like that, but I would like to have it terminate to a router remotely if possible.
Typically Ciso recommends SIP trunk (assuming PSTN trunk) connection to CUBE (voice GW router with CUBE licenses) rather than direct connection to CUCM for many reason, such as demarcation, security, authentication, header modification, etc.
On the gateway router, I have a PRI connected at the present time. I have another location with a CME - SIP PSTN connection. When I went to setup 4 digit dialing between CUCM and CME, I configured the Trunk in the CUCM and had to connect the CME to the CUCM directly. When I attempted to configure the CME to attach to the CUCM gateway router, I get fast busy "unknown number" on phone and when I debug "debug voice ccapi inout" it shows disconnect cause 1.
The site that is currently a CME, will eventually become a gateway on the CUCM. Since the only way I could get the SIP trunk to work was to tie it directly to the CUCM, I thought it was just not possible. So, how do I configure the router to recieve a SIP trunk for the CUCM? I have read 4 different forum posts here that document how to create SIP trunk directly to the CUCM, but none that specify how to terminate it to the gateway router.
That is what I already have, and that works fine. What I want to know is how do it create a call-flow as follows:
ip phone --> cme --> sip-trunk --> Gateway router --> cucm --> ip phone
This is because eventually, the cme will become a gateway router for the cucm, and the cme currently has sip trunk from the PSTN terminating to it at the remote site. With cme, it is very easy to recieve a SIP trunk from the PSTN. How do I get the PSTN sip trunk to route calls in to the gateway router (like a PRI), without directly terminating the PSTN sip trunk to the cucm?
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