search for: rtp

...a SPA2000. I have Asterisk, the ATA186 and Sipura all hard coded to G711ulaw. When the fax machine trains, I get a stream of errors in the log file. The log is attached below. If you have any suggestions they would be GREATLY appreciated. Thank you!!! Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 56 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 94 received Jan 21 16:06:59 NOTICE[10010661]: Unknown RTP codec 60 received Jan 21 16:06:59 NOTICE[10...

...ed with eyebeam, the other one is registered with a SNOM phone. When using the eyebeam client DMTF detection works fine, when using the SNOM phone many digits are missing in the DTMF detection. I analyzed with wireshark and both phones uses RFC 2833 and the trace looks pretty the same. Also the rtp debug log looks fine (see below). What could be the reason? thanks klaus trace: I have entered 1234#, but voicemail received as secret just 123. Got RTP packet from 83.136.33.3:64118 (type 00, seq 042765, ts 4066332168, len 000160) Got RTP packet from 83.136.33.3:64118 (type 00, seq 0...

...cro which answers, 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 wa...

hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556...

Hi, im using a Asterisk Server which is not behind NAT. First i had problems with the fax detection. But this is now solved after adding a wait(2) at the correct place. But i'm still unable to receive a fax due to res_rtp_asterisk.c:3548 ast_rtp_read: RTP Read too short after the Fax session has started. My sip.conf includes [general] allowguest=no alwaysauthreject=yes sendrpid=rpid trustrpid=yes language=de videosupport=yes callevents=yes qualify=yes nat=force_rport,comedia faxdetect=yes t38pt_udptl=yes,redund...

Hi All; I am facing a problem that always the users confused and connect the telephone line coming from the telephone service provider to the FXS port and cause it to be damaged, specially if the card was 2 fxs and 2 fxo, so they make mistake and connect the line to fxs while it should be connected to fxo. What is the solution for this disaster? Regards Bilal

...acing two problems. Problem 1: I got 100 users registered to asterisk from each winsip and then initiates 100 calls from one winsip other winsip. But the problem is approx of 60 calls get mature and asterisk give error for the remaining like shown below. > May 18 14:57:15] WARNING[8314]: rtp.c:2433 rtp_socket: Unable to allocate RTP socket: Too many open files > [May 18 14:57:15] WARNING[8314]: chan_sip.c:6710 sip_alloc: Unable to create RTP audio session: Too many open files > [May 18 14:57:15] ERROR[8314]: acl.c:481 ast_ouraddrfor: Cannot create socket > [May 18 14:57:15] ER...

...y, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the following error : * [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module loaded, can't setup SRTP session. This is a working internal call : > == Using SIP RTP CoS mark 5 > -- Executing [301 at local:1] Dial("SIP/dbucher-00000000", > "SIP/phone1") in new stack > == Using SIP RTP CoS mark 5 > -- Cal...

...h264 (H.264 Video) haddock8-astrx*CLI> *CLI Output:-* -- Executing [111 at bhati-test:1] Answer("SIP/2218-00000664", "") in new stack -- Executing [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_rt...

Hello, What does it mean when "rtp set debug ip" shows RTP packets that have been send, but there is no audio ? There was no audio on my call in both directions, but "rtp set debug" shows that there were RTP packets send. There is no firewall active on my Asterisk server : [root at sip asterisk]# /sbin/service ip...

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[117371...

Audio packets are running... 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28402, Time=73280 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28403, Time=73440 963 16.190381989 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28404, Time=73600...

...add the following to sip.conf of an Asterisk 1.2 system (current production machine/Asterisk as root): tos=0xB8 (Hex B8 = Decimal 184 = Binary 10111000) or if you are running Asterisk v1.4 or newer: tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. tos_video=af41 ; Sets TOS for RTP video packets. To match the current 1.2 machine would I set the Polycom's sip.cfg to the first or second QOS option? Option 1: ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ <QOS> <Ethernet>...

...et debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN IP4 10.2.152.36 a=rtcp:9 IN IP4 0.0.0.0 a=candidate:3607370648 1 udp 2122260223 10.2.152.36 52421 typ host generation 0 network-id 1 network-cost 10 a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host tcptype active generation 0 network-id 1 network-cost 10 bad call...

Hi, 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'...

...STAT_SES_COMPLETE -- 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]...

...domain, but are failing to list trusted domain objects. The wbinfo -u and wbinfo -g commands work fine and list the objects of the "home" domain, but nothing for the trusted domain. The wbinfo -m will not work and produces the following error within the /var/log/samba/winbindd.log file: RTP+vanderce RTP+Vandivl RTP+Vandivlt RTP+vangorr RTP+VEACHJL RTP+Villiaem RTP+Voat RTP+Vogeljs RTP+Wagnerwl RTP+Walkerjv RTP+Wardensd RTP+Watkinrm RTP+Wayh RTP+Weathett RTP+Wedekise RTP+Weekscn RTP+Weissbj RTP+Westsk RTP+Wheelekt RTP+whitakja RTP+Whitela RTP+Whitesje RTP+Wiedmamm RTP+wigginrs RTP+wilk...

...partment for almost a month with no results. I have noticed that I get a message "RFC3389 support incomplete. Turn off on client if possible" in asterisk. I have researched this and made the change in ipmid.cfg (see below), but I am still getting this RFC error. --- ipmid.cfg ---- <RTP qos.ethernet.rtp.user_priority="5"/> <RTP qos.ip.rtp.min_delay="0" qos.ip.rtp.max_throughput="0" qos.ip.rtp.max_reliability="0" qos.ip.rtp.min_cost="0" qos.ip.rtp.precedence="0"/> &l...

...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_rtp_read: Unknown RTP codec 122 received Jun 26 17:56:28 NOTICE[14332]: rtp...

...(preferred 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...