search for: ast_codec_choose

Let's say I have two devices configured and the follow call scenarios occur. [100] disallow=all allow=g722&ulaw Polycom phone with g722,ulaw,alaw,g729 [101] disallow=all allow=ulaw Polycom phone with g722,ulaw,alaw,g729 101 dials 100 -> ulaw to ulaw is chosen 100 dials 101 -> g722 to ulaw is chosen Ideally when 100 dials 101 ulaw would be chosen since it is the common format.

...EBUG[5941] chan_sip.c: *** Our native formats are 0x100 (g729) [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: *** Joint capabilities are 0x100 (g729) [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: *** Our capabilities are 0x10e (gsm|ulaw|alaw|g729) [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x100 (g729) ... 010-12-23 11:06:22] DEBUG[5941] chan_sip.c: build_route: Contact hop: <sip:0010101 at 78.84.207.114:5060;transport=UDP;ob> [2010-12-23 11:06:22] DEBUG[5941] chan_sip.c: Incoming INVITE with 'timer' option supported and "Session-Expires" header....

...ff [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** Our native formats are 0x4 (ulaw) [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** Joint capabilities are 0x0 (nothing) [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** Our capabilities are 0x44 (ulaw|slin) [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: *** Our preferred formats from the incoming channel are 0x40 (slin) [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: This channel will not be able to handle video. [Jul 12 19:08:58] DEBUG[11552] chan_sip.c: Outgoing Call for 5101234567 [Ju...

...[C-000473c5] chan_sip.c: *** Our native formats are (alaw) [Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: *** Joint capabilities are (alaw) [Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: *** Our capabilities are (alaw|ulaw|gsm) [Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: *** AST_CODEC_CHOOSE formats are alaw [Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: *** Our preferred formats from the incoming channel are (alaw) [Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: This channel will not be able to handle video. [Jan 6 11:38:29] DEBUG[5383][C-000473c5] channel_internal_api.c...

I'm attempting to find where my extra long DTMF Tones are coming from. I'm dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway. I'm pressing 4 to select a menu and everything is fine. [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin '4' received on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin passthrough

...;2' (In use) but we don't care because they're not a member of any queue. 13:37:40 chan_sip.c: *** Our native formats are 0x4 (ulaw) 13:37:40 chan_sip.c: *** Joint capabilities are 0xc (ulaw|alaw) 13:37:40 chan_sip.c: *** Our capabilities are 0xe (gsm|ulaw|alaw) 13:37:40 chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x4 (ulaw) 13:37:40 chan_sip.c: This channel will not be able to handle video. 13:37:40 chan_sip.c: build_route: Contact hop: <sip:1000 at 192.168.0.84:27928> 13:37:40 chan_sip.c: SIP/1000-0021a568: New call is still down.... Trying... 13:37:40 chan_sip.c: Trying to put 'SIP/...

Hello you mean while placing a video call ? What info am I looking for in the debug output ? Kind regards. J. On 21-04-17 12:28, Marcelo Terres wrote: > Did you try to activate DEBUG and set the verbosity to a higher level > (100?) to check what Asterisk tells you about? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at

I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regi?es: (11)4063-6100