User at remote site tries to transfer a call from the PSTN to another site.
Call arrives at remote phone, but no audio in either direction. (No re-order or anything, just dead air)
Call information appears on destination phone. (Call appears to be active)
Conference works fine, problem is only with transfer.
Gateway is using MGCP, G.729 inter region.
Local DSPs are used for transcoding and they are registered in CCM
I am going to have the customer switch the codec to G.711 and see if the issue remains. I am leaning towards a transcoding issue, but usually this would result in a fast busy. I have verified the voice path between endpoints and to the CCM.
How did you verified the audio path? And did you mean from two phone to CCM?
You 've said the call flow is call from PSTN to IP Phone. Then IP Phone transfer the call to another IP Phone. So the Audio path should be between PSTN gw to remote Phone. You can use "show voip rtp connection" on gw to check the rtp path. Also you can browse to IP Phone, then select "Stream 1" from the bottom left corner. That should tell you the current rtp path to/from IP Phone.
Wireshark capture from the GW or IP Phone after transfer would be great
Do you experience the following error on Cisco Unity Connection: Sometimes when you try to play or upload Audio Files for greetings in Unity Connection under the Call Handlers such as the System Call Handler Opening Greeting, in Standard/Closed/Holi...
When an Ad hoc conference is triggered by an endpoint to escalate a point-to-point call into a three-way call.HQ-CUCM sends an API request via HTTPS to the Cisco Meeting Server to set up a conference. You can see from the event log of CMS that API us...
You want in some scenarios different URIs or call-IDs and different access code are used by guest and host participants to join a meeting, and different rights or privileges, for example add or remove participants, mute audio or video, or when there...
Opus is an adaptive codec that provides better audio call quality than G.711/G.729 voice codecs and in a low bandwidth environment. The codec has a very low algorithm delay and is it is highly scalable in terms of audio bandwidth, bitrate, and comple...