I didn't know how else to caption this.
I'm trying to play around with codec pass-through. I have two SIP phones, both with g729, behind two Asterisk servers.
I set all the configs, SIP and IAX, to "disallow=all; allow=g729" on both servers.
But the originating server won't even try to call the destination server:
-- Executing Dial("SIP/btel-c7d7", "IAX2/bris/10101") in new stack
Mar 5 02:55:32 WARNING[2786]: channel.c:1942 ast_request: No translator path exists for channel type IAX2 (native 63508) to 256
Mar 5 02:55:32 NOTICE[2786]: app_dial.c:936 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("SIP/btel-c7d7", "") in new stack
== Spawn extension (home, 55, 2) exited non-zero on 'SIP/btel-c7d7'
When I show the peer entries on both servers, I see these same values for the "codec" strings on either end, but they are *different* for the IAX peer than the SIP, e.g. here's a snippet from "show peer:"
iax2 show peer bris * Name : bris Secret : <Set> <other stuff omitted> Codecs : 0xf900 (g729) Codec Order : (g729)
sip show peer btel * Name : btel Secret : <Set> <ditto> Codecs : 0x100 (g729) Codec Order : (g729)
**********************
I'm running CVS-HEAD from yesterday.
I get the same result in reverse if I start the call on the other side.
I have run the Wiki and list archives route; followed the advice there to a tee (add some lines to the general context in sip.con) but nothing seems to yield anything different than the result shown above.
Thanks.
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