Have recently created SIP Trunks as a means of an intercluster trunk to a third party telephone system.
Have created Route Pattern, Route List and Route Group which contains 2 SIP Trunks. Also created New Non Secure SIP Trunk Profile for the SIP Trunks. SIP Trunks using the Standard SIP Profile
My telephone is a 7965 using SCCP.
Route pattern 443X created to use SIP Trunk using standard port of 5060.
Have checked partitions and calling search spaces and these look correct.
Problem is that I receive engaged tone when dialling any 4-digit 443X extension.
CUCM version 7.1.3
Screenshot of SIP Trunk attached.
Thanks for any assistance.
Solved! Go to Solution.
What is the remote PBX
What about the incoming calls from this destination
If the phones are into partition then for the incoming parameteres into the sip trunk(css )
Try to uncheck teh MTP into teh trunks
Finally enable details traces into the cucm do the call and collect the traces form RTMT
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If you are to keep Require MTP on the Trunk, I have found in the past I have needed to actually specify the MRGL on the trunk and rely on the Trunk referencing the MRGL via the Device Pool.
In your case ensure you have a G711alaw mtp resource available or a Transcoder available depending on the Codec used on the third-party PABX.
But like Chrysostomos mentioned, if you enable SIP traces via the serviceability page, then try the call a couple of times. Collect the traces and post them up.
Thanks for the info so far. [and to Chrysostomos1980]
Sorry but I have been trying to track down the relevant file in RTMT but without success, for the last couple of days. I have made a number of calls to the range of extensions over the last couple of days so there will be multiple records but cannot find the relevant log file.