search for: silencesupp

...eplaces Contact: <sip:917070 at 172.16.3.2> Content-Type: application/sdp Content-Length: 258 v=0 o=root 1622 1622 IN IP4 172.16.3.2 s=session c=IN IP4 172.16.3.2 t=0 0 m=audio 12388 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 The following was observed on the 7940's telnet console: SIP Phone> Warning: Unrecognized attribute (silenceSupp) Warning: Unrecognized attribute (silenceSupp) sip_sm_ccb_match_branch_cseq: Method index not found SIPTaskProcessSIPMessage: Error: si...

Hi, I have 2 systems (A and B). I have an 800 number... when someone calls the 800 number it goes: IAX2-->A---IAX---B--->SIP PHONE However.. if the user calling the 800 number is a SIP user that is registered to A it goes: SIP--->A---IAX---B--->SIP PHONE This is the problem... when a call comes in from the IAX2 800 provider, things work fine... however if a SIP user registered to

...IBE, NOTIFY Contact: <sip:9990@172.30.42.5> Content-Type: application/sdp ontent-Length: 235 v=0 o=root 5641 5641 IN IP4 172.30.42.5 s=session c=IN IP4 172.30.42.5 t=0 0 m=audio 29816 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Executing VoiceMailMain("SIP/eden-1000a-4150cc98", "1000@eden") in new stack -- Playing 'vm-password' (language 'en') pbx*CLI> <-- SIP read from 10.253.4.50:5060: INVITE sip:9990@hostname.company.domain;user=phone SIP/2.0 Via...

...replaces Contact: <sip:Voicemail at 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: &l...

...erisk Parkinglot : Use Reason : 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=pt...

...o=root 13636 13636 IN IP4 202.180.nnn.nnn s=session c=IN IP4 202.180.nnn.nnn t=0 0 m=audio 36274 RTP/AVP 18 97 3 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- (15 headers 15 lines) --- Using INVITE request as basis request - 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn Sending to 147.202.nnn.nnn : 5060 (non-NAT) Found peer 'DLS' Found RTP audio format 18 Found RTP audio format 97 Found RTP audio format 3 Found RTP audio form...

...w: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 242 v=0 o=root 2251 2251 IN IP4 24.XX.XXX.101 s=session c=IN IP4 24.XX.XXX.101 t=0 0 m=audio 15202 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 - - - - (no NAT) to 209.XXX.XXX.113:5060 -- Called sip-out/1512484XXX2 Retransmitting #1 (no NAT): INVITE sip:1512484XXX2@209.XXX.XXX.113 SIP/2.0 Via: SIP/2.0/UDP 24.XX.XXX.101:5060;branch=z9hG4bK6adcaf4f From: "512538XXX" <sip:512538XXX@24.XX.XXX.101>;tag=as5329d8fe To: &l...

...IFY Supported: replaces Content-Type: 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: 1...

...IFY Supported: replaces Content-Type: 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: 1...

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

...d: replaces Contact: <sip:99676446 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 a...

...replaces Contact: <sip:Voicemail at 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...

...rted: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 21520 21520 IN IP4 151.196.61.115 s=session c=IN IP4 <public IP> t=0 0 m=audio 11968 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv -- Called bw_outbound/+18885551212 FreePBX*CLI> <--- SIP read from 216.82.224.202:5060 ---> SIP/2.0 100 Giving a try Via: SIP/2.0/UDP public IP:5060;branch=z9hG4bK6ea30a1a;rport=5060 From: "8881231234" <sip:+14105601717 at public IP>;t...

...laces, timer. Content-Type: application/sdp. Content-Length: 279. . v=0. o=root 1631923320 1631923320 IN IP4 192.168.2.10. s=EXAMPLE Systems. c=IN IP4 192.168.2.10. t=0 0. m=audio 52034 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. U 2014/08/12 07:34:20.903830 192.168.2.10:5060 -> 192.168.2.20:5080 INVITE sip:873359633037 at 192.168.2.20:5080 SIP/2.0. Via: SIP/2.0/UDP 192.168.2.10:5060;branch=z9hG4bK541d5594;rport. Max-Forwards: 70. From: "555955599" <sip:555955599 at vic...

...tion/sdp Content-Length: 207 v=0 o=root 5963 5963 IN IP4 192.168.0.203 s=session c=IN IP4 192.168.0.203 t=0 0 m=audio 17456 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - to 192.168.0.153:5060...

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

...E, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 242 v=0 o=root 10295 10295 IN IP4 111.111.111.8 s=session c=IN IP4 111.111.111.8 t=0 0 m=audio 12742 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kasterisk*CLI> Retransmitting #1 (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.11...

...IFY Supported: replaces Content-Type: 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: 1...

...ribute (a): rtpmap:8 PCMA/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 0 PCMU/8000 Media Attribute (a): silenceSupp:off - - - - Media Attribute Fieldname: silenceSupp Media Attribute Value: off - - - - ***************************************************************** So it can be clearly seen how GSM is before G729. Anybody knows if this is an existing bug? Or am I doing something wro...

...d: replaces, timer Contact: <sip:6615xxxxx at 130.117.xxx.xxx> Content-Type: application/sdp Content-Length: 235 v=0 o=root 1698171141 1698171142 IN IP4 130.117.xxx.xxx s=Asterisk PBX 1.6.1.18 c=IN IP4 130.117.xxx.xxx t=0 0 m=audio 39124 RTP/AVP 18 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/800902-00001794 and SIP/130.117.110.21-00001795 >>> ATA ACK's the OK message: <--- SIP read from UDP://82.158.83.xxx:5062 ---> ACK sip:6615xxxxx at 130.117.xxx.xxx SIP/2.0 Via: SIP/2.0/U...