search for: ast_rtcp_read

...ct:1] Set("SIP/03070741-088bd470", "CALLERID(number)=96170707070") in new stack ??? -- Executing [9613070741 at direct:2] Dial("SIP/03070741-088bd470", "SIP/usa/9613070741") in new stack ??? -- Called usa/9613070741 [Jan 28 18:17:36] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short [Jan 28 18:17:42] WARNING[29269]: rtp.c:883 ast_rtcp_read: RTCP Read too short ??? -- Call on SIP/usa-08906450 left from hold ??? -- SIP/usa-08906450 is making progress passing it to SIP/03070741-088bd470 ??? -- SIP/usa-08906450 is ringing ??? -- Call on SIP/usa-08906450 left f...

...! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! Warning, flexibel rate not heavily tested! May 16 06:01:25 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: Received packet with bad UDP checksum May 16 06:01:45 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: Received packet with bad UDP checksum May 16 06:02:05 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: Received packet with bad UDP checksum ........(snip)..... May 16 06:12:05 NOTICE[5627]: rtp.c:355...

Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machine....When running asterisk -rvvvvv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[19803]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:02] WARNING[29804]: rtp.c:891 ast_rtcp_read: RTCP Read too short Can you let me know how to fix this issu...

...terisk console: localhost*CLI> localhost*CLI> -- Executing [ca1@manu:1] Dial("SIP/ua1-ac750040", "SIP/ca1||wWtTkKr") in new stack -- Called ca1 -- SIP/ca1-ab110040 is ringing -- SIP/ca1-ab110040 answered SIP/ua1-ac750040 [Apr 19 16:14:12] WARNING[22989]: rtp.c:874 ast_rtcp_read: RTCP Read too short -- Feature Found: nway-conf-start exten: nway-conf-start -- Executing [s@macro-nway-conf-start:1] Set("SIP/ua1-ac750040", "CONFNO=300") in new stack -- Executing [s@macro-nway-conf-start:2] ChannelRedirect("SIP/ua1-ac750040", "SIP/ca1-ab...

...sk.c:2165 ast_rtp_update_source: Setting the marker bit due to a source update [Jan 30 21:00:01] DEBUG[7119]: taskprocessor.c:484 tps_taskprocessor_destroy: destroying taskprocessor '423a711c-02c7-4b54-ab39-33e6c64e32c3' [Jan 30 21:00:01] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:3284 ast_rtcp_read: Got RTCP report of 76 bytes [Jan 30 21:00:02] DEBUG[7118][C-00000000]: res_rtp_asterisk.c:3284 ast_rtcp_read: Got RTCP report of 76 bytes [Jan 30 21:00:05] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:3284 ast_rtcp_read: Got RTCP report of 76 bytes [Jan 30 21:00:07] DEBUG[7118][C-00000000]: res_r...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

...nel.c:3710 set_format: Set channel DAHDI/65-1 to write format ulaw -- <DAHDI/65-1> Playing 'pbx-transfer.ulaw' (language 'es') [Aug 18 09:16:58] DEBUG[4756]: channel.c:3710 set_format: Set channel DAHDI/65-1 to write format slin [Aug 18 09:16:59] DEBUG[3245]: rtp.c:1246 ast_rtcp_read: Got RTCP report of 108 bytes [Aug 18 09:17:03] DEBUG[3245]: rtp.c:1246 ast_rtcp_read: Got RTCP report of 108 bytes [b] -- Unable to find extension '' in context 'from-pstn' [/b] Please let me to know if you need configuration files. Thanks in advance. Gustavo Duarte.

...s expected. The outgoing message is heard ("The person ... is busy. Please leave a message <beep>"), then is disconnected immediately after the beep. I see the following in the logs: channel.c:2992 in set_format: Set channel Mobile/SCP2500-7fc8 to write format slin rtp.c:1089 in ast_rtcp_read: Got RTCP report of 104 bytes app.c:657 in __ast_play_and_record: One waitfor failed, trying another app.c:661 in __ast_play_and_record: No audio available on Mobile/SCP2500-7fc8?? Calling via any other method to leave voicemail works correctly. Also, when making outgoing calls (eg softphone -&g...

...7 DEBUG[-1260983376]: chan_sip.c:1824 sip_answer: sip_answer (SIP/2000-e12c) Aug 26 15:54:17 DEBUG[-1233335376]: chan_sip.c:817 __sip_ack: Stopping retransmission on '95AB5805-C94F-4C15-AC5A-6DFE5F58D644@192.168.1.101' of Response 46614: Found Aug 26 15:54:17 DEBUG[-1260983376]: rtp.c:378 ast_rtcp_read: Got RTCP report of 84 bytes Aug 26 15:54:17 DEBUG[-1260983376]: rtp.c:378 ast_rtcp_read: Got RTCP report of 118 bytes -- Channel 0/17, span 1 got hangup Urgent handler Aug 26 15:54:17 DEBUG[-1260983376]: channel.c:2559 ast_channel_bridge: Bridge stops because we're zombie or need a soft...

...ing. Unsurprisingly it halted Asterisk and after we cancelled it, it had generated a 1GB file. Both memory logs hinted us to some problem in the RTP module, specifically in the RTCP handling. After investigating the code, looks like for some reason, the res_rtp_asterisk.c (line 4116) when doing the ast_rtcp_read() ended up entering in a loop, causing lots of allocations in the rtp_engine.c (line 2047 - ast_rtp_publish_rtcp_message). Also apparently, another loop (or maybe the same, not sure...) happened calling ast_channel_publish_blob() because we got lots of create_channel_blob_message and ast_channel_sn...

...ommand 'Logoff' == Manager 'admin' logged off from 127.0.0.1 BUT, if I press #8 in the softphone, I can hear the two digit and inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli output in this case is: [Oct 2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' r...

...mand 'Logoff' == Manager 'admin' logged off from 127.0.0.1 BUT, if I press #8 in the softphone, I can hear the two digit and inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli output in this case is: [Oct 2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' r...

...mand 'Logoff' == Manager 'admin' logged off from 127.0.0.1 BUT, if I press #8 in the softphone, I can hear the two digit and inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli output in this case is: [Oct 2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' r...

...mand 'Logoff' == Manager 'admin' logged off from 127.0.0.1 BUT, if I press #8 in the softphone, I can hear the two digit and inmediately the Transfer menu begins playing 'pbx-transfer.gsm'. And the Cli output in this case is: [Oct 2 11:09:17] DEBUG[29533]: rtp.c:1148 ast_rtcp_read: Got RTCP report of 60 bytes [Oct 2 11:09:20] DEBUG[29533]: rtp.c:958 process_rfc2833: - RTP 2833 Event: 0000000b (len = 4) [Oct 2 11:09:20] DEBUG[29533]: rtp.c:806 send_dtmf: Sending dtmf: 35 (#), at 192.168.0.148 [Oct 2 11:09:20] DTMF[29533]: channel.c:2840 __ast_read: DTMF begin '#' r...

...ds: 70 Date: Tue, 02 Dec 2008 12:30:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Really destroying SIP dialog '7c71f80736bb6506511056ce6c070a23 at XX.XX.XX.XX' Method: OPTIONS [Dec 2 12:30:24] WARNING[6669]: rtp.c:891 ast_rtcp_read: RTCP Read too short Reliably Transmitting (NAT) to 68.62.168.138:5060: OPTIONS sip:sarathb at 68.62.168.138:5060 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK34e06ad8;rport From: "asterisk" <sip:asterisk at XX.XX.XX.XX>;tag=as141b747c To: <sip:sarathb at 68.62.168.138...

Does anybody know why I get this error when using zapateller: WARNING[1209214400]: File rtp.c, Line 327 (ast_rtcp_read): RTP Read error: Resource temporarily unavailable It's scrolls until a sound is recived from the line, then it plays the zapateller tones. /Mike

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

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

...et_process: Immediately destroying 2, having received hangup Tx-Frame Retry[-01] -- OSeqno: 013 ISeqno: 012 Type: IAX Subclass: ACK ================================================ ================================================ hacki-mobile*CLI> [Jul 7 14:42:54] DEBUG[9968]: rtp.c:879 ast_rtcp_read: Got RTCP report of 108 bytes [Jul 7 14:42:54] DEBUG[9968]: rtp.c:626 send_dtmf: Sending dtmf: 42 (*), at 10.241.85.100 [Jul 7 14:42:54] DTMF[9968]: channel.c:2463 __ast_read: DTMF begin '*' received on SIP/6002-08383a78 [Jul 7 14:42:54] DTMF[9968]: channel.c:2473 __ast_read: DTMF beg...

...er at (0 requested / 0 actual) timer ticks per second [May 12 22:47:09] DEBUG[22725]: rtp.c: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. [Ma...