06-28-2016 12:49 PM - edited 03-17-2019 07:23 AM
Hello,
I have a customer who has deployed Skype for Business and integrated it with CUCM 10 using an Audiocodes gateway.
After a bit of work we have managed to get phones on CUCM to be able to call Skype clients and vice versa.
The issue I now have is getting Skype clients to dial out to the PSTN via an ISDN30 cicrcuit connected to a Cisco 2900 router acting as an MGCP gateway.
Looking at CUCM traces I can see the SIP call arrive into CUCM and all the details in the Invite look ok.
CUCM responds with a Trying message then a 404 not found.
If I set up a CUCM phone with the same Device Pool and CSS as the SIP trunk to the Audiocodes gateway then I can dial the PSTN destination without any problems so I do not think it is a CSS issue.
I do not see CUCM sending any MGCP commands to the gateway when the SIP Invite arrives so it looks like a CUCM issue.
I have set the Re-routing CSS on the trunk.
I am a bit stumped on how to go forward and am wondering whether I am missing anything obvious - are there any Service Parameters that are required to allow CUCM to act as a transit network?
Any ideas welcome.
06-28-2016 01:49 PM
Hi James,
My two cents, you can try applying rerouting CSS on SIP trunk which contains the route pattern's pointing to the MGCP GW and reset SIP trunk. If MGCP GW is able to handle other calls fine then it is more likely CSS issue.thanks
06-28-2016 03:04 PM
Hi Deepak,
Thank you for responding. I thought that this might be the issue as well. I set the re-routing CSS on the SIP trunk to one that can dial PSTN numbers but that has not helped. I would expect CUCM to send out MGCP commands to the gateway but I am seeing nothing of this type in the CUCM traces.
I will ask the customer to do a debug isdn q931 on the gateway just to be sure the call is not hitting it.
Any other thoughts are welcome.
06-29-2016 04:45 AM
James,
Any chance you can upload SDL traces from you CUCM so that we can analyze, answer should be in the traces.
Leszek
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