search for: ast_rtp_read

...nswers, receives the fax to a tiff, and then runs a script (mailfax) to convert that to pdf and email it. It all works perfectly except for some errors I am seeing in the console. After it hangs up I get a dozen or so messages in the cli saying,"[May 27 14:21:51] WARNING[26225]: rtp.c:1632 ast_rtp_read: RTP Read too short" Perhaps I shouldn't worry since it works but I am nervous to put it in production with these errors. Also, I am not sure that t38 is actually working and being used here. Is there any way to see this at the asterisk cli or is tcpdump the only way? If anyone can sug...

...uting [111 at bhati-test:2] Dial("SIP/2218-00000664", "SIP/2206,60,r") in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Called 2206 -- SIP/2206-00000665 is ringing -- SIP/2206-00000665 is ringing [Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' [Nov 21 15:57:20] NOTICE[30518]: rtp.c:1811 ast_rtp_read: Unknown RTP codec 126 received from '10.10.11.40' -- S...

I was just on 70minute call (IAX2 -> Internet -> IAX2) and during that time I heard several "pops", or "clicks". Each time it happened, I saw the following message: Aug 19 15:36:36 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Any ideas what causes these, and why they turn in to a "pop", instead of just silence, or a missed portion of audio? Thanks Here are the rest of them: ----------------------------------------- Aug 19 15:36:36 NOTICE[1173711792]: rtp.c:429 as...

...MPLETE -- Channel 'SIP/xxx.xxxxxx.se-08aaf470' fax session '0' is complete, result: 'SUCCESS' (FAX_NO_FAX), error: 'CANCELED', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp.c:1434 ast_rtp_read: RTP Read too short [Apr 23 15:30:32] WARNING[31839]: rtp...

...codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered SIP/2007-e4d8 Jul 9 15:50:27 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples 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[1192...

.... 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_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...

Any one see these? Are they benign, or is some system tuning required to remove them? Can't seem to find a resolution in the archives. If you have a link, it would be appreciated. Jun 2 10:58:58 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 19 received Jun 2 10:58:59 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 72 received Jun 2 10:59:00 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 72 received Jun 2 10:59:01 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 19 received...

...anguage 'en') -- Playing 'vm-INBOX' (language 'en') -- Playing 'vm-messages' (language 'en') -- Playing 'vm-onefor' (language 'en') -- Playing 'vm-INBOX' (language 'en') Jul 26 22:40:39 NOTICE[262160]: rtp.c:487 ast_rtp_read: Unknown RTP codec 72 received -- Playing 'vm-messages' (language 'en') Jul 26 22:40:40 NOTICE[262160]: rtp.c:487 ast_rtp_read: Unknown RTP codec 72 received -- Playing 'vm-opts' (language 'en') -- Playing 'vm-changeto' (language 'en') Jul...

...th 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_rtp_read: Unknown RTP > codec 122 received > Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RT...

...start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following messages start to appear (about 100 of them) Nov 23 16:27:25 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP codec 127 received After that asterisk gets totaly confused: Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read to...

...-- Started music on hold, class 'default', on SIP/6004-5e88 -- SIP/6005-5180 is ringing -- SIP/6005-5180 answered SIP/6004-5e88 -- Stopped music on hold on SIP/6004-5e88 -- Attempting native bridge of SIP/6004-5e88 and SIP/6005-5180 Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received Jul 11 15:26:01 NOTICE[14974]: rtp.c:508 ast_rtp_read: Unknown RTP codec 96 received == Spawn extension (from-sip, 6005, 1) exited non-zero on 'SIP/6004-5e88' --------...

...d 13057671523@209.120.202.94:5060 -- SIP/209.120.202.94:5060-0533 is making progress passing it to SIP/1000-c317 -- SIP/209.120.202.94:5060-0533 answered SIP/1000-c317 -- Attempting native bridge of SIP/1000-c317 and SIP/209.120.202.94:5060-0533 Jun 3 22:56:09 WARNING[28769]: rtp.c:500 ast_rtp_read: Forcing Marker bit, because SSRC has changed However the calls complete correctly. I'm using 1.2.8 asterisk stable release. what does that mean? Thanks, Erick.

...gw3*CLI> -- Executing Dial("SIP/332-552e", "SIP/613@fwd") in new stack -- Called 613@fwd -- SIP/fwd-357f is ringing -- SIP/fwd-357f answered SIP/332-552e -- Attempting native bridge of SIP/332-552e and SIP/fwd-357f Sep 13 11:02:52 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:02:57 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:03:01 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:03:02 NOTICE[245776]: rtp.c:489 ast_rtp_read: Unknown RTP codec 72 received Sep 13 11:...

...;< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Control (4) SUBCLASS: T38/Negotiation Requested (19) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:39] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' << [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/GWCISCO5400O-600bfcc8] [Jul 16 17:50:40] NOTICE[2736]: rtp.c:1739 ast_rtp_read: Unknown RTP codec 96 received from '192.168.3.163' << [ TYPE: Null Frame (5) SUBCLASS: N...

...ringing -- SIP/2202-f2ad is ringing -- SIP/2204-11cd is ringing -- SIP/2205-ce62 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- SIP/2205-ce62 answered Zap/1-1 -- Hungup 'Zap/2-1' Jan 22 15:54:25 WARNING[1259633456]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Jan 22 15:54:28 WARNING[1259633456]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Jan 22 15:54:31 WARNING[1259633456]: rtp.c:375 ast_rtp_read: RTP Read error: Resource temporarily unavailable Jan 22 15:56:48 WARNING[12596...

Hi, i'd like to know more about this issue, i'm always getting this message while in call with anyone from sip to zap or zap to sip. ast_rtp_read: Unknown RTP codec 72 received here is my current setup: client side, x-lite, with the transmit silence to yes, using ulaw,alaw on asterisk server side: sip.conf contain allow=ulaw and allow=alaw dtmfmode=inband So i always get this anoying notice and i cannot find any doc about fixing it. I hav...

...p://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT, Asterisk console (run with asterisk -vvvvvvc) started to show the following messages: 1. All codecs disabled in X-Lite/Pro except iLBC and Speex or only Speex: Sep 27 12:15:45 NOTICE[135169024]: rtp.c:420 ast_rtp_read: RTP: Received packet with bad UDP checksum Sep 27 12:15:45 NOTICE[135169024]: rtp.c:420 ast_rtp_read: RTP: Received packet with bad UDP checksum Sep 27 12:15:45 NOTICE[135169024]: rtp.c:420 ast_rtp_read: RTP: Received packet with bad UDP checksum Sep 27 12:15:45 NOTICE[135169024]: rtp.c:420 ast...

I'm testing NVBackgroundDetect with Sipura-300 and I get this error: rtp.c:505 ast_rtp_read: Unknown RTP codec 100 received21 Does anybody know what is it? -- #Joseph

the above-message keep popping up every second during a conversation between a zap(fxs) channel and sip channel. * eventually hung after a long while we can talk to each other and we can ring one another without any problem. i've had x-lite and x-pro register with * without this problem. furthermore, i have ask my friend to turn off all codec expect g.711MLAW; that did not help i then

...e debug below: Mediatrix forces (*) to use Payload Type as 96: [...] a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [...] Then we've got this nice debug from (*): May 5 10:48:15 NOTICE[139648000]: rtp.c:418 ast_rtp_read: Unknown RTP codec 96 received May 5 10:48:15 NOTICE[139648000]: rtp.c:418 ast_rtp_read: Unknown RTP codec 96 received We had this static_RTP_PT[xx] structure in rtp.c :(asterisk source): [...] [34] = {1, AST_FORMAT_H263}, [97] = {1, AST_FORMAT_ILBC}, [101] = {0, AST_RTP_DTMF}, [110] = {1, AST_FO...