We recently migrated from analog trunks to SIP trunks on our CME system. After doing some testing we see that inbound calls from PSTN to extensions (SCCP 79xx phones), AA and CUE are working except for one peculiar situation as follows:
A PSTN call is answered by a person at a DID and then _transferred_ to another extension. No one answers at that extension therefore the call is CFNA to CUE. The call will be connected but the caller will get dead air. The caller can leave a voicemail and use CUE options but will not hear the prompts. The voicemail will be recorded correctly and the called extension can retrieve it and listen to it. So it appears that call setup works but no RTP from CUE back to PSTN.
All other cases work fine. If a caller gets the AA they will hear prompts, select extensions and leave messages normally. If a caller calls a DID and the called DID CFNA to CUE it will also work normally. Only when a call is forwarded to CUE by another extension do we see this problem.
I have done a bit of troubleshooting and see two things that stick out.
- The recorded message will not show as coming from the PSTN. It will show as coming from the extension that transferred it. Whereas calls directly to DID or AA show the correct PSTN number
- When checking SIP messages a see references to G729 codec when the call connects from the PSTN (G729 is only codec supported by our SIP provider) but in subsequent SIP messages between extension and CUE do not reference any codec at all. I think transcoding is fine because of the other cases where CUE works but maybe the transfer procedure breaks the transcoding?
The attached debug shows a PSTN call from 3055551212 to 8888888888 which is the DID for 206. 206 then transferred to 202 and 202 CFNA to CUE. Extension 202 can then hear the message and it appears to have come from 206. The caller got dead air so he had to speak on faith that the CUE was on the other end and was recording.
Any ideas would be greatly appreciated.