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Is it Possible to Trunk Call Manager to Shoretel?

Has anyone ever trunked Cisco Call Manager to Shoretel? PRI? H323? SIP?

I am looking for some interoperability docs

We have a potential customer running shoretel at several sites, we would like to add new sites on Call Manager 4.2 or possible 7.1 and have the sites communicate to the shoretel sites (4 digit dialing)



I am also having a very similar issue.  I have a SIP trunk to the PSTN on an asterisk box and calls come in fine through the PSTN --> asterisk --> CUCM DN.  If I try to dial out from an IP phone or softphone registered in CUCM through the SIP trunk, I get the message "Your call can not be completed as dialed, check your directory and dial again."  I tried creating a softphone extension in asterisk and I can dial the CUCM DN perfectly, yet even when I created a specific route pattern to dial the one extension in asterisk, I get the same message from CUCM.  I tried the DNA and it shows the call as routing out through the SIP trunk, no issue.  I have tested network connectivity between the systems and it is fine also.  SIP trunk is set to use UDP.  Any suggestions?

Also, I ran wireshark on the connection.  When I made a call from the xlite softphone from asterisk --> CCM, i showed the normal SIP traffic.  When I make a call from IP Communicator --> asterisk, NO SIP traffic appears.


The same suggestions apply to you. 'Cisco Lady' tells users it isn't working; she doesn't tell you what the problem is.

Also just because you can't send a call out of a trunk doesn't mean you have the same issue - you have a different system you are integrating to, and the resolution is likely to be different. You should open a new thread.



Aaron Please remember to rate helpful posts to identify useful responses, and mark 'Answered' if appropriate!
Eric Knudson
Cisco Employee

OLD post, but I recently ran across this and found _no_ useful information on SIP trunking online.

Symptom was that we could receive calls over a SIP trunk to a Shoretel switch but we could not send calls across the SIP trunk (from an 8.6 CUCM).

We determined today that the Shortel switch didn't like an INVITE without the codec SDP, so enabling Early Offer or checking the MTP Required on the SIP trunk fixed the problem.


Same old post & same old question.

We have a new customer who is trying to migrate from Shoretel to Cisco. At present they have Shoretel PBX in US and India. Now as per the 1st phase of the migration they are going to remove the shoretel from india and place a cucm. They will remove shoretel from US after one year only in the 2nd phase. So now we have to some how route the calls between CUCM & Shoretel PBX. Any information on the CUCM & Shoretel PBX integration steps would be a great help!!!

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