search for: process_sdp

...sterisk, not the res_fax and res_fax_digium that comes with FFA. What happens is sometimes the T.38 negotiation goes well and others it fails completely. That's what I got from the debug info on two different calls, without changing any configs: [Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP o=PVG 1265107050040 1265107050040 IN IP4 10.152.0.164... UNSUPPORTED. [Feb 2 08:38:56] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing session-level SDP...

hi, is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ? or is chan_pjsip better supported? or the recommended way for asterisk is use respoke.io? my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia canreinvite=no encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=yes transport=wss,ws dtlsrekey=60 dtlsverify=no dtlscertfile=/etc/pki/tls/certs/rapidssl.c...

...pgrade asterisk server to 1.8.x from my asterisk 1.6, But when making A Call from SIP Client, I got cli Warning ... and no call has been made. My Sip Client is using lib java peers client http://peers.sourceforge.net/ with standard codec PCMU/PCMA [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8 101 [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:9029 process_sdp: Failing due to no acceptable offer found the strange thing is when using asterisk 1.6, is normal, when using asterisk 1.8.x and using another client like Ekiga is normal too,...

Hi, I'm using asterisk 1.2.1. Is there anybody out there who knows what this warning means? *WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38* Google does not help at all. TIA Giorgio Incantalupo

...t this is the message. == Registered translator 'g729tolinb' from format 8 to 6, cost 99999 == Registered translator 'lintog729b' from format 6 to 8, cost 18 == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs! WARNING[5126]: File chan_sip.c, Line 1601 (process_sdp): No compatible codecs! I want to use a snom 100 with this codec. THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/2003051...

Anyone else have issues with T.38 where the call drops after T.38 is attempted to be negotiated, with a message like the below? WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

...- - *CLI> sip show peer andrew [snip] Codecs : G.729A But when we try to use more than one (such as transferring an incoming BRI call to a second phone), when the phone answers, the transfer fails and we get the following: *CLI> Jul 25 16:49:25 WARNING[114695]: chan_sip.c:2820 process_sdp: No compatible codecs! Jul 25 16:49:28 WARNING[114695]: chan_sip.c:2820 process_sdp: No compatible codecs! Jul 25 16:49:28 WARNING[114695]: chan_sip.c:2820 process_sdp: No compatible codecs! -- Executing Dial("SIP/andrew-89e3", "SIP/jeremy|20") in new stack -- Called jer...

...3878 ast_rtp_write: Ooh, format changed from unknown to alaw [May 12 22:47:09] DEBUG[22725]: rtp.c:3904 ast_rtp_write: Created smoother: format: 8 ms: 20 len: 160 [May 12 22:47:13] DEBUG[22725]: rtp.c:1240 ast_rtcp_read: Got RTCP report of 88 bytes [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8223 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8223 process_sdp: Processing session-level SDP o=- 840372135 840372136 IN IP4 125.213.160.145... UNSUPPORTED. [May 12 22:47:15] DEBUG[22587]: chan_sip.c:8223 process_sdp: Processing session-level SDP s=...

...that to me: v=0 o=CSM 0 1 IN IP4 x.x.x.x s=Acme c=IN IP4 x.x.x.x t=0 0 m=audio 22152 RTP/AVP 8 0 18 4 101 a=rtpmap:101 telephone-event/8000 And here's the debugging: [May 8 17:45:30] DEBUG[6444]: chan_sip.c:5092 do_setnat: Setting NAT on RTP to Off [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP o=CSM 0 1 IN IP4 x.x.x.x... UNSUPPORTED. [May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP s=Acme... UNSUPPORTED. [May 8 17...

Hello folks, I'm running 1.8.11 on a Centos 6 system with an adjacent Hylafax server using softmodems: Noticed this in the Asterisk log when trying to send a fax from Hylafax to Asterisk: [Apr 3 01:53:09] WARNING[29184]: chan_sip.c:8926 process_sdp: Multiple audio streams are not supported I've googled a few asterisk tickets that may suggest that yes, multiple audio streams are not supported in 1.8, but is there possibly a way for multiple audio streams to be supported? Thank you very much.

...0 -- Call accepted by 127.0.0.1 (format alaw) -- Format for call is (alaw) -- IAX2/300-7211 is ringing -- IAX2/300-7211 answered SIP/TOOTAiAudio-00008262 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 [2015-02-17 16:52:51] NOTICE[3467][C-00001d5b]: chan_sip.c:10645 process_sdp: T.38 re-INVITE detected but no fax extension [2015-02-17 16:52:56] WARNING[3467][C-00001d5b]: chan_sip.c:9868 process_sdp: Insufficient information for SDP (m= not found) -- Executing [h at from-internal:1] Hangup("SIP/TOOTAi-00008262", "") in new stack == Spawn exten...

....1 (format alaw) > -- Format for call is (alaw) > -- IAX2/300-7211 is ringing > -- IAX2/300-7211 answered SIP/TOOTAiAudio-00008262 > == Using UDPTL TOS bits 184 > == Using UDPTL CoS mark 5 > [2015-02-17 16:52:51] NOTICE[3467][C-00001d5b]: chan_sip.c:10645 > process_sdp: T.38 re-INVITE detected but no fax extension > [2015-02-17 16:52:56] WARNING[3467][C-00001d5b]: chan_sip.c:9868 > process_sdp: Insufficient information for SDP (m= not found) > -- Executing [h at from-internal:1] Hangup("SIP/TOOTAi-00008262", "") > in new stack...

...OK (using alaw codec) 2) Echo functionality on phone 2: OK (using alaw codec) 3) Call from phone2 to phone1: OK (both using alaw) 4) Call from phone1 to phone2: immediate disconnect after answering (might not be related) console says: [Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10434 process_sdp: Received AVP profile in audio answer but AVPF is enabled: audio 7200 RTP/AVP 8 101 [Aug 20 09:19:57] WARNING[7080][C-0000011f]: chan_sip.c:10819 process_sdp: Failing due to no acceptable offer found I enabled debug on the IP of the dect-phone (full log attached), but it does not make me any wi...

When I get a SIP call, I get the following error: *CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 5147771111, 1) == Spawn extension (incoming, 5147771111, 1) exited non...

...ack -- Called [VoIP-provider]/[ext. number dialed] -- SIP/[VoIP-provider]-77a8 is ringing -- SIP/[VoIP-provider]-77a8 answered SIP/[ID of Cisco]-4663 -- Attempting native bridge of SIP/[ID of Cisco]-4663 and SIP/[VoIP-provider]-77a8 2005-07-13 10:20:48 NOTICE[23276]: chan_sip.c:2792 process_sdp: No compatible codecs! 2005-07-13 10:20:48 NOTICE[23280]: channel.c:1736 ast_set_read_format: Unable to find a path from g729 to ulaw 2005-07-13 10:20:48 NOTICE[23280]: channel.c:1703 ast_set_write_format: Unable to find a path from alaw to g729 2005-07-13 10:20:48 WARNING[23280]: chan_sip.c:1836 s...

...t=dynamic context=95040 dtmfmode=rfc2833 disallow=all allow=g729 insecure=port,invite canreinvite=no my extension is : exten => 321,1,Dial(SIP/321) when i want to set up a call (123 dial 321). but failed. it says: == Using SIP RTP CoS mark 5 [Aug 20 18:23:21] NOTICE[14543]: chan_sip.c:8454 process_sdp: No compatible codecs, not accepting this offer! Could you tell me what 's wrong? -- Thanks & Regards Sucan

...register sipura spa2 with 2 lines on asterisk. 2. use first line to call somewhere. 3. while using first line try to call from second line somewhere else in 3 step i hear fast busy tones on second line and asterisk console gives me this short error: Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No compatible codecs! My sipura adapter is using g729a codec. When using both of sipura lines separately everything works fine, until someone tries to use both lines simultaneously. Any advice ? Tofik Suleymanov

...a2501") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/66.193.176.35-000000b8", "sip/ra2501,360") in new stack == Using SIP RTP CoS mark 5 -- Called sip/ra2501 -- SIP/ra2501-000000b9 is ringing [2013-02-24 17:05:12] WARNING[6554]: chan_sip.c:8979 process_sdp: ignoring 'video' media offer because port number is zero -- SIP/ra2501-000000b9 answered SIP/66.193.176.35-000000b8 -- Locally bridging SIP/66.193.176.35-000000b8 and SIP/ra2501-000000b9 [2013-02-24 17:05:31] WARNING[6554]: chan_sip.c:8979 process_sdp: ignoring 'video' me...

...outbound proxy. Kamailio routes signaling to Asterisk, and then back to clients. Currently the problem is RTP, when INVITE is received from client A to Kamailio, it is relayed to Asterisk. Asterisk responds with 488 Not Acceptable here and the cli says: NOTICE[11642][C-00000006]: chan_sip.c:10124 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 30212 RTP/SAVPF 111 103 104 0 8 106 105 13 126 WARNING[11642][C-00000006]: chan_sip.c:10509 process_sdp: Rejecting secure audio stream without encryption details: audio 30212 RTP/SAVPF 111 103 104 0 8 106 105 13 126 S...

Hi all: I recently update a system from CVS (Asterisk CVS-09/25/03-15:58:42), and I receiving the following message: *CLI> WARNING[1187305408]: File chan_sip.c, Line 1864 (process_sdp): No compatible codecs! The "show codecs" command shows: *CLI> show codecs 1 (1 << 0) G.723.1 2 (1 << 1) GSM 4 (1 << 2) G.711 u-law 8 (1 << 3) G.711 A-law 16 (1 << 4) MPEG-2 layer 3 32 (1 << 5) ADPCM 64 (1 << 6) 16 bit Signed Linear PCM 128...