Hi,
I have just signed up with Voxee.com and have attached my Asterisk
server to dial them via IAX2.
Below is the start of the log which dials the number and promply
hangs up when the call is answered, with the logs saying that the
channel is not compatiable.
I have traced this down to the g.729 codec which I don't have
installed. Any ideas on how to force that the codec not be used?
BTW, I have disallow=all and allow only the codecs that I want to use
in both iax.conf and sip.conf.
Best Regards,
Todd Reese
-- Executing SetCallerID("SIP/201-fbb8", "6788896066") in new stack
-- Executing Dial("SIP/201-fbb8",
"IAX2/134:[EMAIL PROTECTED]/17702561571") in new stack
-- Called 134:[EMAIL PROTECTED]/17702561571
-- Call accepted by 66.246.246.52 (format g729)
-- Format for call is g729
Jun 8 18:48:41 NOTICE[6073]: channel.c:1884 set_format: Unable to
find a path from g729 to gsm
Jun 8 18:48:41 NOTICE[6073]: channel.c:1884 set_format: Unable to
find a path from g729 to gsm
Jun 8 18:48:42 NOTICE[6073]: channel.c:1884 set_format: Unable to
find a path from g729 to gsm
Jun 8 18:48:42 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to
transmit frame type 256, while native formats is 2 (read/write = 2/2)
Jun 8 18:48:42 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to
transmit frame type 256, while native formats is 2 (read/write = 2/2)
Jun 8 18:48:42 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to
transmit frame type 256, while native formats is 2 (read/write = 2/2)
............................
Jun 8 18:48:51 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to
transmit frame type 256, while native formats is 2 (read/write = 2/2)
Jun 8 18:48:51 WARNING[6405]: chan_sip.c:2170 sip_write: Asked to
transmit frame type 256, while native formats is 2 (read/write = 2/2)
-- IAX2/66.246.246.52:4569-7 answered SIP/201-fbb8
Jun 8 18:48:51 WARNING[6405]: channel.c:2308
ast_channel_make_compatible: No path to translate from SIP/201-fbb8(2)
to IAX2/66.246.246.52:4569-7(256)
Jun 8 18:48:51 WARNING[6405]: app_dial.c:1324 dial_exec_full: Had to
drop call because I couldn't make SIP/201-fbb8 compatible with
IAX2/66.246.246.52:4569-7
-- Hungup 'IAX2/66.246.246.52:4569-7'
== Spawn extension (local-access, 17702561571, 2) exited non-zero on
'SIP/201-fbb8'
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