An update from a couple weeks ago - After a bit of pushing by me, Dialogic says that they do not currently nor do they plan to support forking which is the cause of the problem for them. As a result, the gateway cannot handle multiple phones ringing on one extension nor the way SIPX voice mail answers an unanswered call. Dah. I mentioned that I know that Audiocodes, Patton, and the phones we have tested all work fine and Dialogic seem to be the only things I tested that fails on this case.
Didn't get too far because they have very few support calls with problems caused by lacking the ability to handle these calls. I have tried Audiocodes and it works fine. Unfortunately, Dialogic has a niche in that it emulates Siemen's equipment to a Siemen's PBX so that the PBX will pass full call information (like caller id) to it. Siemen's switches will refuse to send this information to 3rd party products over other types of connections such as a T1. Their suggested workarounds to try are: 1) Use Asterisk server to route calls between the gateway and SIPX or 2) Try using B2BUA function being built into SIPX v4. The idea is to isolate the Dialogic from the forking. Would be nice if the thing just worked the way it should. Give suggestion #2, Is the functionality of B2BUA in v4 worth looking at? Are things in the development builds to the point were it is worth trying as an experiment. I pretty sure I could get the Asterisk idea to work but really hate to have an Asterisk server around just for that purpose. Tony Wyland Director of Network Services Messiah College [EMAIL PROTECTED] 717-766-2511 x2380 >>> "Scott Lawrence" <[EMAIL PROTECTED]> 9/29/2008 12:25 PM >>> On Mon, 2008-09-29 at 11:26 -0400, Tony Wyland wrote: > 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. That trace shows two 180 responses - the first is from a phone, and the second from the voicemail system (note that they have different values for the 'tag' parameter on the To headers). These each create different early dialogs, and provide different SDP - perfectly normal and legal. The SDP in the 200 from the voicemail system matches what it sent in its 180, which is correct. It does look as though the Dialogic is unhappy, since it sends the BYE less than 2ms after the ACK. I suggest you get logs from the gateway and try to figure out what it objects to. If its problem is that the SDP in the two ringing responses are different (or, put another way - that the SDP in the 200 does not match that in the first 180), then that's a bug in the Dialogic stack. _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
