I have an existing 2921 CME/CUE version 7.x which currently has a PRI to the PSTN. I am trying to switch over to a SIP trunk provided by XO Communications. Right now my DID numbers only match the last two digits of my ephone-dn number. For example my DID would be something like this 330-555-8510 and my ephone-dn would be 1210 I am using a num-exp for each DID number as a translation. So for the example I just gave I have the following: num-exp 3305558510 1210. My CME router is not the main data router for the site's data network, but it is the default gateway for the phones. I have about 30 SCCP phones, and I have a few analog stations that I'm connecting to via pots dial-peers. My plan is to plug the sip trunk, which is being delivered on a seperate /30 WAN segment by the carrier, into one of the unused ethernet ports on the CME and setting a route to the carrier's SIP network which will use that interface. For example, I have a default route pointing to the site's main gateway, and I have a more specific route for the carrier's SIP network pointing to the interface whiere my SIP trunk terminates.
I have added all of my VOIP Dial-peers and shutdown my pots dial-peers that were pointing to the PRI. When I do this and try to make a test call from one of my SCCP phones I get an intercept message from the carrier stating that the device I am using is not registered. If I add a secondary number to the ephone-dn as the full e.164 number of my test number I can make a call out, and I can make a call back in (on my test number). However when I make a call out, the caller ID shows up as unknown (this may be just because there is no association in the carrier's database with my test number...I have not ported my existing numbers over yet).
I would like to know what I need to do so that I don't have to put a secondary dn on all of my ephone-dn's, is there any sort of translation (say, with a sip-profile maybe) that I can do so that when my endpoints attempt to register, the SIP header gets re-written from [email protected] to [email protected] going out, and gets inversely translated coming back in? If I do this with a sip-profile, can somebody share with me an example of how this is configred? There seems to be LOTS of SIP headers that can be re-written with a sip-profile and I need some advice on what to do.
Also, is there anything additional I would need to do for my analog station devices? Currently I just have a station-id configured on the port, and a pots dial-peer sending the destinantion pattern to the port the fax machine is plugged into.
Any thing you can share that would point me in the right direction would be appreciated!!