search for: 000160

...xecuting [41325122774 at from-sip:1] Answer("SIP/sipcall.ch-0000008d", "") in new stack > 0x7f3964080f30 -- Probation passed - setting RTP source address to 123.456.789.123:20600 Got RTP packet from 123.456.789.123:20600 (type 00, seq 042281, ts 1387619622, len 000160) -- Executing [41325122774 at from-sip:2] Set("SIP/sipcall.ch-0000008d", "DB(lastcaller/number)=987654321") in new stack -- Executing [41325122774 at from-sip:3] GotoIf("SIP/sipcall.ch-0000008d", "0?black,1") in new stack -- Executing [41325...

...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 042766, ts 4066332328, len 000160) Got RTP packet from 83.136.33.3:64118 (type 00, seq 042770, ts 4066332968, len 000160) Got RTP packet from 83.136.33.3:64118 (type 00, seq 042771, ts 4066333128, len 000160) Got RTP packet from...

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 iptables status iptables: Firewall not running. Kind

...on 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 but RTP looks like bad call (1.1.1.1 is "public" ip of PSTN SIP GW) Got  RTP packet from    1.1.1.1:13460 (type 08, seq 002433, ts 000160, len 000160) Sent RTP packet to      10.2.152.36:63249 (type 08, seq 022470, ts 000160, len 000160) Got  RTP packet from    1.1.1.1:13460 (type 08, seq 002434, ts 000320, len 000160) Sent RTP packet to      10.2.152.36:63249 (type 08, seq 022471, ts 000320, len 000160) Got  RTP packet from    1...

...01 is ringing > -- SIP/phone1-00000001 is ringing > -- SIP/phone1-00000001 is ringing > -- SIP/phone1-00000001 answered SIP/dbucher-00000000 > -- Remotely bridging SIP/dbucher-00000000 and SIP/phone1-00000001 > Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160) > Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160) > Got RTP packet from 192.168.128.99:49646 (type 126, seq 031575, ts > 000001, len 000000) > [Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk...

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

...g -- SIP/phone1-00000001 is ringing -- SIP/phone1-00000001 is ringing -- SIP/phone1-00000001 answered SIP/dbucher-00000000 -- Remotely bridging SIP/dbucher-00000000 and SIP/phone1-00000001 Sent RTP P2P packet to 192.168.128.99:49646 <http://192.168.128.99:49646> (type 00, len 000160) Sent RTP P2P packet to 192.168.128.99:49646 <http://192.168.128.99:49646> (type 00, len 000160) Sent RTP P2P packet to 192.168.128.99:49646 <http://192.168.128.99:49646> (type 00, len 000160) Got RTP packet from 192.168.128.99:49646 <http://192.168.128.99:49646> (type 126,...

...can see: *CLI> -- Executing [111 at my-phones:1] Answer("SIP/2001-081dd6e0", "") in new stack -- Executing [111 at my-phones:2] VoiceMail("SIP/2001-081dd6e0", "2000") in new stack Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037718, ts 000160, len 000160) -- <SIP/2001-081dd6e0> Playing 'vm-intro' (language 'en') Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037719, ts 000320, len 000160) Sent RTP packet to 58.251.75.228:9956 (type 00, seq 037720, ts 000480, len 000160) Sent RTP packet to 58...

hi, i have following topology PSTN - Asterisk ---- internet -----  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

...ho audio. debian*CLI> -- Executing [222 at my-phones:1] Answer("SIP/2000-b6d06750", "") in new stack -- Executing [222 at my-phones:2] Echo("SIP/2000-b6d06750", "") in new stack Got RTP packet from 192.168.1.102:42406 (type 00, seq 003468, ts 2904300, len 000160) Sent RTP packet to 192.168.1.102:42406 (type 00, seq 002928, ts 2904296, len 000160) Got RTP packet from 192.168.1.102:42406 (type 00, seq 003469, ts 2904460, len 000160) Sent RTP packet to 192.168.1.102:42406 (type 00, seq 002929, ts 2904456, len 000160) Got RTP packet from 192.168.1.102:42406 (t...

...Probation passed - setting RTP source address to 178.119.159.58:44704 [Aug 9 22:15:52] > 0x7fc5dc014060 -- Probation passed - setting RTP source address to 178.119.159.58:44704 [Aug 9 22:17:08] Sent RTP packet to 178.119.146.190:59051 (type 08, seq 028865, ts 2789673216, len 000160) [Aug 9 22:17:09] Sent RTP packet to 178.119.146.190:59051 (type 08, seq 028866, ts 2789673376, len 000160) [Aug 9 22:17:09] Sent RTP packet to 178.119.146.190:59051 (type 08, seq 028867, ts 2789673536, len 000160) [Aug 9 22:17:09] Sent RTP packet to 178.119.146.190:59051 (type...

...9 typ host > tcptype active generation 0 network-id 1 network-cost 10 > > > but RTP looks like > > bad call (1.1.1.1 is "public" ip of PSTN SIP GW) > > Got  RTP packet from 1.1.1.1:13460 <http://1.1.1.1:13460> (type > 08, seq 002433, ts 000160, len 000160) > Sent RTP packet to 10.2.152.36:63249 <http://10.2.152.36:63249> > (type 08, seq 022470, ts 000160, len 000160) > Got  RTP packet from 1.1.1.1:13460 <http://1.1.1.1:13460> (type > 08, seq 002434, ts 000320, len 000160) > Sent RTP packet t...

...hm=MD5 [Aug 11 15:53:50] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 [Aug 11 15:53:50] Organization: Doubango Telecom RTP debug : RTP Debugging Enabled for address: 178.119.146.190:0 [Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type 08, seq 014114, ts 3292374327, len 000160) [Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50760 (via ICE) (type 08, seq 033787, ts 3292374320, len 000160) [Aug 11 16:01:20] Got RTP packet from 178.119.146.190:58814 (type 08, seq 014115, ts 3292374487, len 000160) [Aug 11 16:01:20] Sent RTP packet to 178.119.146.190:50...

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ?? Kind regards. On 10-08-16 21:53, Matt Fredrickson wrote: > I don't see an ice-ufrag or

...102 ACK User-Agent: PortaOne Max-Forwards: 69 Content-Length: 0 <-------------> [2020-09-19 23:42:22] VERBOSE[2637] chan_sip.c: --- (12 headers 0 lines) --- [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020640, ts 000160, len 000160) [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020641, ts 000320, len 000160) [2020-09-19 23:42:22] VERBOSE[15153][C-00021a1f] res_rtp_asterisk.c: Sent RTP packet to 41.11.11.11:13918 (type 8, seq 020642,...

Hi, Use orecx, voip call recording and monitoring. www.orecx.com Thanks & Regards, Vidura Senadeera, Sri Lanka. Tel - +94114520001 Mobile - +94777766596 yahoo/skype Ids - vidurased > ------------------------------ > > Message: 17 > Date: Fri, 30 Nov 2007 08:58:41 +0530 > From: ram <talk2ram at gmail.com> > Subject: Re: [asterisk-users] Newb Question > To:

...on SIP/patton-0000002b == Using SIP RTP CoS mark 5 == Extension Changed 249[subs] new state Ringing for Notify User 749 == Extension Changed 249[subs] new state Ringing for Notify User 750 -- SIP/249-0000002c is ringing Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043637, ts 000160, len 000160) Sent RTP packet to 192.168.102.200:5030 (type 00, seq 043638, ts 000320, len 000160) Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101014/505d8b9c/attachment.htm

Hi, I have Asterisk set up on Fedora with a single SIP trunk, with a few handsets configured. The Asterisk box has both public and private addressing, so "canreinvite=no" is set on both the SIP trunk and handset configurations so I can get around the nasty NAT issues. One odd behaviour I am seeing is certain destinations are resulting in different SIP codes being sent back to Asterisk,

...on 02080) Got RTP packet from x.x.x.x:22760 (type 101, seq 042363, ts 175520, len 000004) Got RTP RFC2833 from x.x.x.x:22760 (type 101, seq 042363, ts 175520, len 000004, mark 0, event 00000006, end 1, duration 02080) Got RTP packet from x.x.x.x:22760 (type 00, seq 042364, ts 176480, len 000160) Got RTP packet from x.x.x.x:22760 (type 00, seq 042365, ts 176640, len 000160) Got RTP packet from x.x.x.x:22760 (type 00, seq 042366, ts 176800, len 000160) Got RTP packet from x.x.x.x:22760 (type 00, seq 042367, ts 176960, len 000160) <SIP/STARMG1-000003c5>AGI Rx << STREA...

...d the paging by itself works great. However, when I try it with the audio file, it starts to play correctly, then abruptly hangs up after 6 or 7 seconds. When I turn debug on, this is what I see: [2015-03-19 15:46:38.292] Sent RTP packet to X.X.X.X:1049 (type 00, seq 037511, ts 061440, len 000160) [2015-03-19 15:46:38.299] Got RTP packet from X.X.X.X:1049 (type 00, seq 036666, ts 4175299232, len 000160) [2015-03-19 15:46:38.312] Sent RTP packet to X.X.X.X:1049 (type 00, seq 037512, ts 061600, len 000160) [2015-03-19 15:46:38.316] Got RTP packet from X.X.X.X:1049 (type 00, se...