I have heard from others having similar issue with Dialogic. A transfer via the AA does work; the only failure is when the phone rings and then attempts to go to voice mail due to no answer.
I didn't have a chance to do a trace on the SipX side yet as suggested by Scott but I do have a packet trace showing the gateway doing a SIP BYE right after the OK from SipX. Attached is also a text graph done by Wireshark on this call. Tony Wyland [EMAIL PROTECTED] >>> "Tony Graziano" <[EMAIL PROTECTED]> 9/25/2008 5:20 PM >>> Without a call trace it's hard to tell, normally I think that maybe the DIALOGIC has a problem with REFER or hasn't been told explicitly how to handle it. I've seen that with gateways. What would be a good litmus test is to see if the AA works or if it drops the calls after the AA says "please hold while i transfer your call". The call would have to traverse the gateway though, an internal call is not a good test, except to say it does work, but not through the gateway. This would be the same as a transfer to voicemail after the ringtimeout has been reached. When the AA transfers a call, it is a REFER. So is forwarding to voicemail. The are other gateways that handle this without issue (Patton for instance). Check the DIALOGIC manual for REFER. >>> "Tony Wyland" <[EMAIL PROTECTED]> 09/25/08 16:44 PM >>> Curious if anybody has had success with Dialogic series DMG gateways. We are experimenting with one and have one issue that seems likely to be caused by the gateway. First, what works --- calls to SipX extensions work fine, calls that go redirect to voicemail because there is no registered phone work fine, calls from SIP phones going back to the gateway work fine. Transfers work fine. Lots of things work except for one case - If a call comes from the Dialogic to SipX and a registered phone rings then the call goes to voicemail because the phone was not answered, the call is dropped by the Dialogic. When the gateway sees a request to connect to a different media than was indicated in the "ringing" messages, it seems to loose its mind a bit. It also forgets to tell the PBX to disconnect the call. While working with Dialogic on the case, I wanted to see if anybody had seen and fixed this problem before. Perhaps some magic parameter would make it feel better. Another indicator of a Dialogic issue is that calls between SipX registered phone are always fine as well as calls through an Audiocodes gateway. Tony Wyland Director of Network Services Messiah College [EMAIL PROTECTED] 717-766-2511 x2380 _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
vm08-no.pcap
Description: Binary data
|Time | 153.42.16.234 | 153.42.16.172 | |0.000 | INVITE SDP ( g711U g711A telephone-event CN t38) |SIP From: sip:[EMAIL PROTECTED]:5060 To:sip:[EMAIL PROTECTED] | |(1232) ------------------> (5060) | |0.005 | 100 Trying| |SIP Status | |(1232) <------------------ (5060) | |0.514 | 180 Ringing |SIP Status | |(1232) <------------------ (5060) | |16.400 | 180 Ringing |SIP Status | |(1232) <------------------ (5060) | |16.407 | 200 OK SDP ( t38 g711U g711A telephone-event) |SIP Status | |(1232) <------------------ (5060) | |16.472 | RTP (g711U) |RTP Num packets:2 Duration:0.019s SSRC:0x16DB1244 | |(49094) <------------------ (9000) | |16.496 | ACK | |SIP Request | |(1052) ------------------> (5060) | |16.498 | BYE | |SIP Request | |(1052) ------------------> (5060) | |16.515 | 200 OK | |SIP Status | |(5060) <------------------ (5060) |
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