creating conference to 2 pstn numbers via h323 gateway

Paul Austin · ‎08-14-2013

Hi we have a cucm system (v8.X) with a sip trunk to a SIP pbx and H323 gateway to the PSTN. All works fine regarding calls in/out between PSTN and both the cucm and SIP PBX. However if we receive an external call to an extension on the SIP PBX and then try to transfer it to another PSTN number, the call fails with fast busy tone when the far end answers. we have G711 codec throughout and have tried HW conference and transcoder resources local on the gateway with no luck.

Any Ideas? Will changing the gateway to MGCP help?

Thanks All.

Chris Deren · ‎08-14-2013

What is the PSTN connection type, i.e. PRI, FXO, etc?

Chris

Paul Austin · ‎08-14-2013

The PSTN connectivity is Primary rate

Chris Deren · ‎08-14-2013

OK, post "debug isdn q931" of the failed call.

Chris

Paul Austin · ‎08-14-2013

will need to be a while as I'm in the UK and its nearly the middle of the night.

Paul Austin · ‎08-16-2013

There is no failed call - I just cannot transfer the call from the SIP PBX to the 2 PSTN calls (1 received and 1 made). I have also made the gateway MGCP with no sucess either.

Amine Nouasri · ‎08-14-2013

Are both CUCM and your SIP PBX sharing the same voice gateway? if it is the case, I would do first a debug isdn q931 to see if the call is leaving the voice gateway. This is to narrow down your issue to the VG or something in the CUCM - SIP PBX side.

Paul Austin · ‎08-14-2013

The PSTN gateway recieves the initial call then gets routed by the callmanger down the SIP trunk to the PBX, then the SIP PBX initiates a transfer, dials the PSTN number that is processed by the callmanger and send the call out via the same PSTN gateway.

hope that makes it clearer.

Chris Deren · ‎08-16-2013

OK, so what happens to the transferred call? If it does not go anywhere that means it fails. What do you hear/see?

Chris