FYI - bug 1043 has been fixed on Feb 18:
"From my log, below, you will see that ast_rtp_bridge is not comparing the codecs properly. Asterisk is currently comparing the integers, and not the bits of the codec.
In the below example codec0 = 260. That means Codec0 allows both 256 (g729) and 4 (uLaw). So that would mean that Codec0 and Codec1 do have a "Codec Match".
Asterisk needs to do a bit compare, and not a int compare in this case."
-- SIP/dialnet-8bac answered SIP/chris0-df00 -- Attempting native bridge of SIP/chris0-df00 and SIP/dialnet-8bac Feb 16 11:27:48 WARNING[68889520]: rtp.c:1204 ast_rtp_bridge: codec0 = 260 is not codec1 = 256, cannot native bridge. Feb 16 11:27:50 NOTICE[68889520]: rtp.c:264 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Feb 16 11:27:50 NOTICE[68889520]: rtp.c:293 process_rfc3389: Don't know how to handle RFC3389 for receive codec 256
>>
I have the same problem with codec negotiation, my Voip provider use g729 however I have also connection with Iaxtel which only use GSM. I can only get one or the other codec working when dialing out.
My iax.conf setting is below: ; Inter-Asterisk eXchange driver definition
[general] port=4569 bandwidth=low disallow=all allow=gsm allow=g729 disallow=g723.1 ; Hm... Proprietary, don't use it... disallow=lpc10 ; Icky sound quality... Mr. Roboto. jitterbuffer=yes dropcount=3 maxjitterbuffer=250 maxexccessbuffer=50 register => dkwok:[EMAIL PROTECTED]
tos=lowdelay [iax_home] type=friend context=int-ext auth=md5 user=iax_home secret=cccccc trunking=yes disallow=all allow=gsm host=dynamic qualify=yes
[iaxtel] type=friend disallow=all disallow=g729 allow=gsm trunking=yes context=from-iaxtel [atp] type=friend disallow=all allow=g729 trunking=yes context=atp host=xxx.xxx.xxx.xxx
I would like to hear any comment from * developer.
-- David Kwok
Iaxtel/FWD # 17001813482 ext 1002
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