search for: sendrecv

...TE request body from SIP carrier) v=0 o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46 s=sip call c=IN IP4 38.126.208.46 t=0 0 m=audio 30552 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 (inside 200 OK response body from asterisk) v=0 o=root 835643920 835643920 IN IP4 201.234.196.171 s=Asterisk PBX 11.10.0 c=IN IP4 201.234.196.171 t=0 0 m=audio 12112 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20...

...with Android based setup video?mail is not stored (audio is through). Both the client?use H.264 codec with following sdp information: Android Based Client SDP Parameters v=0 o=- 1325786904 1325786904 IN IP4 172.16.130.47 s=Polycom RealPresence c=IN IP4 172.16.130.47 b=AS:1920 t=0 0 a=sendrecv m=audio 3230 RTP/AVP 118 115 114 113 0 8 119 a=rtpmap:118 SIRENLPR/48000 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/800...

...sion: 'MGCP 0.1' 3 headers, 0 lines Handling request 'NTFY' on aaln/S0/SU0/0@MRP200-S1 Transmitting: 200 13 OK to 10.0.2.6:2427 -- Endpoint 'aaln/S0/SU0/0@MRP200-S1-1' observed 'hd' -- Creating connection for aaln/S0/SU0/0@MRP200-S1-1 in cxmode: sendrecv callid: 6b49143d0fa02584 We're at 10.0.2.2 port 19844 Answering with capability 4 Posting Request: CRCX 7 aaln/S0/SU0/0@MRP200-S1 MGCP 1.0 C: 6b49143d0fa02584 L: p:20, a:PCMU M: sendrecv X: 0fa02584 v=0 o=root 2307 2307 IN IP4 10.0.2.2 s=session c=IN IP4 10.0.2.2 t=0...

...ession is simply dial into 600 demo extension - echo test ... Handling request 'NTFY' on aaln/1@10.0.1.19 Transmitting: 200 29 OK to 10.0.1.19:2427 -- Endpoint 'aaln/1@10.0.1.19-1' observed '0' -- MGCP Asked to indicate tone: on aaln/1@10.0.1.19-1 in cxmode: sendrecv Posting Request: RQNT 306 aaln/1@10.0.1.19 MGCP 1.0 X: 2149c6df R: hu(N), hf(N), D/[0-9#*](N) to 10.0.1.19:2427 -- MGCP asked to indicate -1 'UNKNOWN' condition on channel MGCP/aaln/1@10.0.1.19-1 -- MGCP Asked to indicate tone: on aaln/1@10.0.1.19-1 in cxmode: sendrecv Posti...

....37.86.203:2427Verb: '510', Identifier: '15', Endpoint: 'Protocol', Version: 'Error or' 1 headers, 0 lines -- MGCP asked to indicate -1 'UNKNOWN' condition on channel MGCP/aaln/0@ubr924-1 -- MGCP Asked to indicate tone: on aaln/0@ubr924-1 in cxmode: sendrecv Posting Request: RQNT 16 aaln/0@ubr924 MGCP 1.0 X: 0cd7cc1f R: hu(N), hf(N), D/[0-9#*](N) to 65.37.86.203:2427 -- MGCP Asked to indicate tone: cg on aaln/0@ubr924-1 in cxmode: sendrecv Posting Request: RQNT 17 aaln/0@ubr924 MGCP 1.0 X: 0cd7cc1f R: hu(N), hf(N), D/[0-9#*](N) S: cg to 65.37.86...

Hello, I have just received an MGCP Phone for test purpose and I can't place a call from my MGCP Phone. I can call my MGCP phone from a SIP Phone. Here is my mgcp.conf: ; ; MGCP Configuration for Asterisk ; [general] ;port = 2427 ;bindaddr = 0.0.0.0 ;[dlinkgw] ;host = 192.168.0.64 ;context = default ;line => aaln/2 ;line => aaln/1 [192.168.10.10] host = 192.168.10.10 context =

...using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our Vendor sees it: Here's what I see. 1. The first call is initiated. (CRCX) The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately. All other CAs that we've worked with do not instruct us to go to sendrecv mode until the number has been completely dialed. 2. The call is terminated when hung up. The call agent responds to this, but it never tells us to delete th...

...4 192.168.50.10 b=CT:1920 t=0 0 m=audio 48260 RTP/AVP 100 101 9 8 0 102 b=TIAS:64000 a=rtpmap:100 G7221/16000 a=fmtp:100 bitrate=32000 a=rtpmap:101 G7221/16000 a=fmtp:101 bitrate=24000 a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:102 telephone-event/8000 a=fmtp:102 0-15 a=sendrecv m=video 48262 RTP/AVP 97 98 99 34 31 c=IN IP4 192.168.50.10 b=TIAS:1920000 a=rtpmap:97 H264-RCDO/90000 a=fmtp:97 profile-level-id=008016;max- mbps=42000;max-fs=3600;max-smbps=323500 a=rtpmap:98 H264/90000 a=fmtp:98 profile-level-id=428016;max-mbps=35000;max-fs=3600;max-smbps=323500 a=rtpmap:99 H263...

...=IN IP4 192.168.2.1 t=0 0 m=audio 7070 RTP/AVP 8 0 3 97 18 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <--- Transmitting SIP response (733 bytes) to UDP:217.10.79.9:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 V...

...t 10.2.0.2> Content-Type: application/sdp Content-Length: 256 v=0 o=root 27452 27452 IN IP4 10.2.0.2 s=session c=IN IP4 10.2.0.2 t=0 0 m=audio 13256 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> Retransmitting #1 (no NAT) to 10.2.0.203:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK2343a87a;received=10.2.0.203 From: "Jim Felderman" <sip:1101 at 10.2.0.2>;tag=001d45b61d4906959ea33ab4-af2b7b8b To: <sip:Voicemail at 10.2.0.2>;ta...

...NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 261305180 261305180 IN IP4 192.168.254.44 s=- c=IN IP4 192.168.254.44 t=0 0 m=audio 16392 RTP/AVP 18 100 101 a=rtpmap:18 G729a/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv INVITE sip:203@192.168.121.20 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.44:5060;branch=z9hG4bK-3fda1e15 From: PAP 220 <sip:220@192.168.121.20>;tag=b8b86be991749af5o0 To: <sip:203@192.168.121.20> Call-ID: a44265f9-c09c6825@192.168.254.44 CSeq: 101 INVITE Max-Forwards: 70 Contact: PAP 220 &lt...

...tion/sdp Content-Length: 412 v=0 o=root 3831 3831 IN IP4 216.82.224.202 s=session c=IN IP4 216.82.224.202 t=0 0 m=audio 17050 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 12426 RTP/AVP 31 34 103 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=sendrecv _________________________________________________________________ Windows Live: Make it easier for your friends to see what you?re up to on Facebook. http://windowslive.com/Camp...

...: No Encryption : No When the call comes in the SDP contains :- v=0. o=root 973184584 973184584 IN IP4 81.x.x.x s=session. c=IN IP4 81.x.x.x t=0 0. m=audio 11370 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. and we reply back with :- v=0. o=root 822402971 822402971 IN IP4 88.x.x.x s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.x t=0 0. m=audio 10428 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. When we send t...

...application/sdp Content-Length: 287 v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk*CLI> <--- SIP read from 10.0.0.10:5060 ---> SIP/2.0 486 Busy Here From: "Shawn Ip"<sip:user at 192.168.10.2>;tag=as25dd7670 To: <sip:5551212 at 10.0.0.10> Call-ID: 0a72cba50ae922e206a7f65c69bc4670 at 192.168.10.2 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168...

...application/sdp Content-Length: 287 v=0 o=root 3386 3386 IN IP4 192.168.10.2 s=session c=IN IP4 192.168.10.2 t=0 0 m=audio 17492 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- asterisk*CLI> <--- SIP read from 10.0.0.10:5060 ---> SIP/2.0 486 Busy Here From: "Shawn Ip"<sip:user at 192.168.10.2>;tag=as25dd7670 To: <sip:5551212 at 10.0.0.10> Call-ID: 0a72cba50ae922e206a7f65c69bc4670 at 192.168.10.2 CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168...

im having a strange problem related to call-limit for peers. well im not sure if its related to call-limmit or not. Bottom line is: I call a user A, from user B. user B hears silence, untill it goes to voicemail. when user B hangsup. user B's call limit is reset to 0 but user A's call limit is not reset.strange thing is user A's status on cli is shown as NOANSWER, while user B did not

...at 10.0.0.8> Content-Type: application/sdp Content-Length: 254 v=0 o=root 3609 3609 IN IP4 10.0.0.8 s=session c=IN IP4 10.0.0.8 t=0 0 m=audio 14398 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ===================================================== ================ext to ext=============================== SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK280378de3608b5bf1.2576ced30c6198b74;received=10.0.0.46 From: "ext" <sip:117 at 10.0.0.8>;tag=d729237fcc To:...

Dan Cropp wrote: > I had my screenshots flipped. Is there a way to make sure the Contact field is NOT included in the ACK response to the OK (for the Answer)? > > PJSIP is including the Contact for the ACK response to the OK. > Contact:<sip:1234 at xxx.xxx.xx.xxx:5060> > There is no configuration option to configure this behavior. What is the full SIP signaling? -- Joshua

...t 10.2.0.2> Content-Type: application/sdp Content-Length: 256 v=0 o=root 27452 27452 IN IP4 10.2.0.2 s=session c=IN IP4 10.2.0.2 t=0 0 m=audio 12088 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [Jan 19 14:33:01] NOTICE[15644]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms? Sending to 10.2.0.203 : 5060 (no NAT) cworks-phones1*CLI> <--- Transmitting (no NAT) to 10.2.0.203:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.2.0.203:5060;branch=z9hG4bK8a2...

...304 v=0 o=102 3079422269 3079422292 IN IP4 10.44.1.20 s=X-Lite c=IN IP4 10.44.1.20 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Attempting SIP protocol listen on: 10.44.1.20:5060 Established SIP protocol listen on: 10.44.1.20:5060 SEND TIME: 3079423989 SEND >> 0.0.0.100:5060 INVITE sip:100 SIP/2.0 Via: SIP/2.0/UDP 10.44.1.20:5060;rport;branch=z9hG4bK42C9872FBBB23E82A267FF9CA39FD454 From: informatica <sip...