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:XXXXXXXX@66.246.246.52/17702561571") in new stack -- Called 134:XXXXXXX@66.246.246.52/17702561571 -- 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'
Kanuri, Seshu (Company IT)
2005-Jun-09 06:40 UTC
[Asterisk-Users] format g729 and Voxee.com
Voxee will not accept any calls that are not in G729. You need G729 codec on your Asterisk. Period. Seshu -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rich Adamson Sent: Thursday, June 09, 2005 10:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; Todd Reese Subject: Re: [Asterisk-Users] format g729 and Voxee.com> 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-------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited.
> 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:XXXXXXXX@66.246.246.52/17702561571") in new stack > -- Called 134:XXXXXXX@66.246.246.52/17702561571 > -- 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'The above implies that Voxee.com is configured for g729 only. I don't use this itsp, but you might check their web site or call them to see what codec options are available. The flip side is go order and install g729 from digium.