No audio if call goes through Unity Connection

[apu]

We have a Cisco 2911 router with connections to two ITSPs as we are provisioning the new circuit before porting all the numbers from the old provider. On our network, we have CUCM and Unity Connection servers, both running 10.5.2.

Outbound via both providers works without a problem. If a call comes in from the old provider, whether it rings directly to a Cisco IP phone (via a translation pattern) or goes to the auto-attendant (also via a translation pattern) and then you dial an extension, there are no issues. Similarly, if a call comes in from the new provider, if it rings directly to a IP phone, there is no problem.

However, if the inbound call from the new provider first goes to Unity Connection, and you dial an extension, you get no audio (in either direction) after "please wait while I transfer your call." The call does ring to the IP phone so the DTMF is fine but neither party can hear the other.

Any ideas?

Thanks, Apu

[Mike_Brezicky]

What codec are you using?
What s your connection setup - SIP/H323?
You might need to invoke an MTP or transcoder resource. You can test this out by forcing an MTP on the inbound SIP Trunk / H323 Gateway.

[apu]

g711ulaw for all calls, both to/from the provider and as reported by the IP phone, SIP connections to both ITSPs.

[Scott Leport]

Hi,

Can you please post the configuration of your gateway / CUBE?

Have you also checked that the MRGL in CUCM contains an annunciator?

We may require some debugs from gateway / CUBE displaying a working and a non-working call to assist further.

[JH51928]

Hi APU, did you make it work now?

I'm having the same issue right now, no audio when call is transferred from the call handler, whether putting an extension number or by the caller inputs. We didn't make any changes to the config, it was working fine for three years or so just suddenly won't work properly.

Could I ask if anyone has solution to fix this?

[M Reza]

hi friends,

i had this issue too..

after "Check" MTP option (Media Termination Point) on Sip-Trunk between CUCM and Router-Cube,

audio is working properly and fine..

[Mohamed H. Abdelmotaleb]

1- are you using SIP Circuit from the new provider or E1/T1 or Analog lines FXO ?

2- please paste dial-peers configuration from router.

3- make sure SDP message from CUCM to CUC is using codec G711ulaw

4- take debug CCSIP messages from voice gateway and call manager logs from CUCM and check those logs at Translator X