search for: ilbctolin_framein

...I try to set up planet VIP-050 with asterisk. Everything works fine instead of the call transfer. When I press # console says something like this: >Oct 5 11:11:20 DEBUG[25104]: chan_sip.c:2222 sip_rtp_read: Oooh, format changed >to 1024 >Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP (4)? >Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_framein: Huh? An ilbc >frame that isn't a multipleof 50 bytes long from RTP (4)? >Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144 ilbctolin_fra...

...les for format G726 Jul 9 15:50:27 NOTICE[1192491824]: channel.c:1508 ast_set_read_format: Unable to find a path from G726 to SLINR Jul 9 15:50:27 NOTICE[1192491824]: channel.c:1478 ast_set_write_format: Unable to find a path from ILBC to G726 Jul 9 15:50:27 WARNING[1192491824]: codec_ilbc.c:141 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (80)? Jul 9 15:50:27 WARNING[1192491824]: chan_sip.c:1333 sip_write: Asked to transmit frame type 1024, while native formats is 16 (read/write = 64/1024) -- Hungup 'Zap/1-1' == Spawn extension (from-sip, 0221700...

i tried asterisk ilbc codec against kphone. when the call got connected, i started to immediately get these kind of message to the console: WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)? WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)? WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An il...

...P SIGNED MESSAGE----- Hash: SHA1 Hi, I'm setting up an Asterisk system which is connected to an Alcatel 4400 PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a call by hitting the # key, I get this messages and nothing happens on the phone: WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? But I'm not allowing ilbc as codec? If I prevent * from loading ilbc I tet this error and the call is hung up, when transfering: Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to find a codec tran...

Hi I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also tried inband) and I get the following error: june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? This means that I cannot get access to voicemail from the handsets !!! Any clues??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/...

...les for format G726 Jul 9 13:15:37 NOTICE[1192491824]: channel.c:1508 ast_set_read_format: Unable to find a path from G726 to SLINR Jul 9 13:15:37 NOTICE[1192491824]: channel.c:1478 ast_set_write_format: Unable to find a path from ILBC to G726 Jul 9 13:15:37 WARNING[1192491824]: codec_ilbc.c:141 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (40)? Jul 9 13:15:37 WARNING[1192491824]: chan_sip.c:1333 sip_write: Asked to transmit frame type 1024, while native formats is 16 (read/write = 64/1024) I will appreciate any help. Kind regards, Miguel

...061-92e0 and SIP/10.10.10.10-78fa == Spawn extension (iaxtest, 2144466715, 3) exited non-zero on 'SIP/53061-92e0' The call drops. If I enable ILBC codec with Asterisk, here is what I get: == Forcing Marker bit, because SSRC has changed Jun 26 17:56:28 WARNING[14332]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (160)? Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rt...

...== Spawn extension (iaxtest, 2144466715, 3) exited non-zero on > 'SIP/53061-92e0' > > The call drops. > > If I enable ILBC codec with Asterisk, here is what I get: > > == Forcing Marker bit, because SSRC has changed > Jun 26 17:56:28 WARNING[14332]: codec_ilbc.c:175 ilbctolin_framein: > Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP > (160)? > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP > codec 122 received > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP > codec 122 received > Jun 26 17:56...

Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers