Mark Henry wrote: > > 1. Your softphone is not sending g729 This was a SIP trace of a successful u-law call. In an earlier post Kamlesh provided a trace of a failed G.729 call which did not include the dialog between the Asterisk server and the ITSP. I asked for this trace so that I could see the codecs offered by the ITSP.
> 2. canreinvite should be set to yes for using pass-thru mode I believe that by pass-thru mode [1] Kamlesh means he wants to avoid transcoding from G.729 to another codec since that requires a license per channel. Pass- thru mode can be achieved with "canreinvite=no" as shown by the following line from the successful u-law SIP trace and Mark Michelson's asterisk-dev post: >> [Jun 3 13:11:32] -- Packet2Packet bridging SIP/100-000034d8 and >> SIP/yyy.yyy.yyy.yyy-000034d9 >From [asterisk-dev] Native Bridging: terminology [2]: ...within SIP, native bridging has two subcategories. One, typically referred to as "SIP native bridging" is used when reINVITEs are enabled. The endpoints send their media directly to one another. The other subcategory is called "Packet 2 Packet" or "P2P" bridging. If reINVITEs are not enabled, but there are also no features that require the Asterisk core to be in the voice path, then the bridging will be done at the RTP layer of Asterisk. [1] http://www.voip-info.org/wiki/view/Asterisk+G.729+pass-thru [2] http://lists.digium.com/pipermail/asterisk-dev/2010-March/043053.html Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
