search for: set_format

...h on 'Zap/1-1' -- Executing Dial("Zap/1-1","IAX2/kumara@teliax/01194777070239|30|tr") in new stack -- Called kumara@teliax/01194777070239 -- Call accepted by 208.139.204.228 (format g729) -- Format for call is g729 Jun 18 19:17:28 NOTICE[8554]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 18 19:17:28 NOTICE[8554]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 18 19:17:28 NOTICE[8554]: channel.c:1884 set_format: Unable to find a path from g729 to gsm -- IAX2/teliax-2 is ringing -- Nobody picked up in 30000 ms...

...ll. It rings on the other side, and of course as I don't have any g729 licenses installed it drops the call when answered. -- Executing Dial("SIP/amejia-1fc8", "H323/######@H323gw") in new stack -- Called #######@H323gw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19 15:22:43 WARNING[75484]: channel.c:2323 set_format: Unable to find a codec translation path from g729 to ulaw Apr 19...

...nswer("H323/ip$172.z.z.z:4836/14", "") in new stack -- Executing Dial("H323/ip$172.z.z.z:4836/14", "SIP/user@193.y.y.y") in new stack -- Called user@193.y.y.y -- SIP/user@193.y.y.y-40455d68 is ringing Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec translation path from g729 to ulaw Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec ........ ....... translation path from g729 to slin Dec 15 14:45:13 WARNING[19795]: channel.c:2380 set_format: Unable to find a codec translation path from g7...

Hi list! My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a voicemail. On two of these numbers the voicemail works without any problem, on the other it doesn't... I get this error: [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) [Oct 17 17:01:29] WARNING[14700]: file.c:957 ast_streamfile: Unable to open /var/spool/asterisk/voicemail/default/00390151111111/unavail (format 0x100 (g729)): No such file or directory [Oct 17 17:01:29] WARNING[14700]: channel...

...t;) in new stack -- Goto (sky,s,1) -- Executing [s at sky:1] Playback("Skype/rexesbposolutions-084159e8", "enter") in new stack -- <Skype/rexesbposolutions-084159e8> Playing 'enter' (language 'en') [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x4 (ulaw) [Dec 29 17:27:14] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore format back to 4 -- Executing [s at sky:2] Queue("Skype/rexesbposolutions-084159e8", "markq|t|||900") in new stack -- St...

...it from teliax for dialing through them. Third, I have somewhat poor support from teliax since I have send them 3,4 emails and so far i got no replies. Please, help me to go ahead from this point Thanking all of you, Kumara. The error I got Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a path from g729 to gsm Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a path from g729 to gsm -- IAX2/teliax-1 is ringing -- IAX2/teliax-1 answered Zap/1...

...oh323.conf I get an error about trying to use ulaw: -- Executing Dial("SIP/D0103-3a43", "OH323/80152392522|50") -- H.323 call to 80152392522 with codec(s) g729 -- Called 80152392522 -- OH323/80152392522-ef38 is ringing Jun 3 10:09:05 NOTICE[27744]: channel.c:1873 set_format: Unable to find a path from ulaw to g729 Jun 3 10:09:05 NOTICE[27744]: channel.c:1873 set_format: Unable to find a path from ulaw to g729 OH323/80152392522-ef38: Format changed to ulaw (native g729). Jun 3 10:09:05 NOTICE[27744]: channel.c:1873 set_format: Unable to find a path from g729 to ul...

...k -- 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...

...: ernestine ip monitor failure for dozer ping Context: outbound-swift Whenever I try to copy this callfile into /var/spool/asterisk/outgoing/ I get these 3 lines repeating over and over (I'm not 100% sure which entry is first): [2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353 set_format: Unable to find a codec translation path from (nothing) to (slin) [2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: file.c:1017 ast_streamfile: Unable to open AAA/check_ip_failure (format (nothing)): Function not implemented [2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: app_playback.c:484 playbac...

set_format: Unable to find a codec translation path from ulaw to g729 Both endpoints are PAP2 set to G711 only I have 1.2.17 on Suse 10.1

...729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) -- Format for call is g729 Apr 24 15:38:38 NOTICE[5586]: channel.c:1833 set_format: Unable to find a path from g729 to slin . . . . I get ringback from Nufone, but as soon as the call answers I get an error: Apr 24 15:43:42 NOTICE[5596]: channel.c:1833 set_format: Unable to find a path from g729 to slin . . . What am I doing wrong to cause it to want to transcode? I assum...

...I put G711 and no silence suppression. In asterisk I have some errors : [Feb 14 11:28:55] WARNING[10547]: channel.c:3033 ast_channel_make_compatible: No path to translate from SIP/sip_trunk_gva-mg-02-006f37f0(256) to SIP/0625037998-006de430(8) [Feb 14 11:29:04] WARNING[10546]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 14 11:29:04] WARNING[10546]: channel.c:2702 set_format: Unable to find a codec translation path from alaw to g729 [Feb 14 11:29:05] NOTICE[10547]: rtp.c:772 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). P...

...;) in new stack [Aug 4 17:50:11] DEBUG[17094]: pbx.c:3692 pbx_extension_helper: Launching 'SayDigits' [Aug 4 17:50:11] -- Executing [s at connect-to-me:2] SayDigits("SIP/gafachi1a-00000000", "'1'") in new stack [Aug 4 17:50:11] DEBUG[17094]: channel.c:3881 set_format: Set channel SIP/gafachi1a-00000000 to write format slin [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3878 ast_rtp_write: Ooh, format changed from unknown to ulaw [Aug 4 17:50:11] DEBUG[17094]: rtp.c:3904 ast_rtp_write: Created smoother: format: 4 ms: 20 len: 160 [Aug 4 17:50:11] DEBUG[17094]: channel.c...

...ed on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format to g729 from alaw native formats 0x8 (alaw) [Jul 15 18:59:55] WARNING[8752]: channel.c:5254 set_format: Unable to find a codec translation path from 0x8 (alaw) to 0x100 (g729) [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: Asked to transmit frame type g729, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: A...

...1 answered Jun 6 21:08:42 DEBUG[2121]: channel.c:777 channel_find_locked: Avoiding initial deadlock for 'Zap/1-1' -- Zap/1-1 answered Zap/3-1 Jun 6 21:08:42 DEBUG[2134]: chan_zap.c:4817 zt_indicate: Requested indication -1 on channel Zap/3-1 Jun 6 21:08:42 DEBUG[2134]: channel.c:2348 set_format: Set channel Zap/3-1 to read format slin Jun 6 21:08:42 DEBUG[2134]: channel.c:2348 set_format: Set channel Zap/1-1 to write format slin Jun 6 21:08:42 DEBUG[2134]: channel.c:2348 set_format: Set channel Zap/1-1 to read format slin Jun 6 21:08:42 DEBUG[2134]: channel.c:2348 set_format: Set chann...

....0 ; Address to bind to context = default ; Default for incoming calls of not registered phones disallow = all allow = g729 allow = g723 allow = alaw allow = ulaw Connected a 'Sipura SPA' sip phone to asterisk with g729 as its preferred codec. Problem: asterisk cannot make the connection: set_format: Unable to find a codec translation path from alaw to g729 Connection only can be made if the Sipura's preferred codec is alaw/ulaw. Any help appreciated Tamas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/att...

...e. When the message is supposed to be recorded, the voicemail app bombs and this is displayed on the console: -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/105/INBOX/msg0001 format: g729, 0x8140f88 Jun 1 10:08:45 WARNING[15148]: channel.c:2326 set_format: Unable to find a codec translation path from g729 to slin Jun 1 10:08:45 WARNING[15148]: app.c:621 ast_play_and_record: Unable to set to linear mode, giving up Obviously I don't have codec_g729 installed. The real question is, why does it need to convert to slinear? Thanks! -- Kristia...

...in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to find a codec translation path from ilbc to ulaw Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to find a codec translation path from ilbc to ulaw Jun 26 17:53:52 WARNING[14248]: chan_sip.c:2555 sip_write: Asked to transmit frame type 1024, while native formats is 4 (re...

I am having a problem with one of my callers who is using either g729 or alaw.  I can do alaw but not g729 so asterisk should negotiate alaw right?  In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up.  Why? Last minute thought: Is it possible that the caller is sending g729 in RTP even though the SIP negotiation clearly chooses alaw? Maybe I need some RTP debugging. Asterisk 13.14.1 on Debian, using c...