Re: [Freeswitch-users] SIP 183 on a leg before bleg invite in bridge
Hi Guys I have a situation where a call is received on a-leg, FS sends 183 session progress to the a-leg before the call is attempted in b-leg. Then, when the b-leg attempts the call for the bridge and sends an invite, it receives 180 ringing, and this is not relayed back through the a leg, so the a-leg never hears ringing. The call goes: Cisco UA (1.1.1.1) -> FS (2.2.2.2) -> Gateway (3.3.3.3) Sofia config has nothing special in it, relevent sections: <param name="inbound-codec-negotiation" value="generous"/> <param name="disable-transcoding" value="true"/> <param name="manual-redirect" value="true"/> <param name="disable-transfer" value="true"/> and the dial plan is straight forward, just a bridge between the two endpoints. Any ideas? Here is the trace of the call: INVITE sip:1234@2.2.2.2:5060 SIP/2.0. Date: Fri, 20 Apr 2012 07:58:36 GMT. Call-Info: <sip:1.1.1.1:5060>;method="NOTIFY;Event=telephone-event;Duration=500". Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY. From: "User" <sip:00000000@1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567. Allow-Events: presence, kpml. Supported: 100rel,timer,resource-priority,replaces. Supported: X-cisco-srtp-fallback. Supported: Geolocation. Min-SE: 1800. Content-Length: 212. To: <sip:1234@2.2.2.2>. Expires: 180. Content-Type: application/sdp. Call-ID: a088a380-f911172c-3d33e6-8d200795@1.1.1.1. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979. CSeq: 101 INVITE. Session-Expires: 1800. Max-Forwards: 69. . v=0. o=CiscoSystemsCCM-SIP 2000 1 IN IP4 1.1.1.1. s=SIP Call. c=IN IP4 1.1.1.1. t=0 0. m=audio 31422 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. U 2012/04/20 08:58:16.142711 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979. From: "User" <sip:00000000@1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567. To: <sip:1234@2.2.2.2>. CSeq: 101 INVITE. User-Agent: FreeSWITCH. Content-Length: 0. . U 2012/04/20 08:58:16.169453 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979. From: "User" <sip:00000000@1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567. To: <sip:1234@2.2.2.2>;tag=vm45yDZjgmDKH. Call-ID: a088a380-f911172c-3d33e6-8d200795@1.1.1.1. CSeq: 101 INVITE. Contact: <sip:1234@2.2.2.2:5060;transport=udp>. User-Agent: FreeSWITCH. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 247. Remote-Party-ID: "1234" <sip:1234@2.2.2.2>;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1334881876 1334881877 IN IP4 2.2.2.2. s=FreeSWITCH. c=IN IP4 2.2.2.2. t=0 0. m=audio 26820 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. U 2012/04/20 08:58:16.176126 2.2.2.2:5080 -> 3.3.3.3:5060 INVITE sip:05600660128@3.3.3.3:5060 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5080;rport;branch=z9hG4bK44BS1jH9Za6yS. Max-Forwards: 68. From: "00000000" <sip:00000000@2.2.2.2>;tag=4vjXj69rtpaeS. To: <sip:1234@3.3.3.3:5060>. Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b. CSeq: 27118284 INVITE. Contact: <sip:gw+C4DF4B2A088C91@2.2.2.2:5080;transport=udp;gw=C4DF4B2A088C91>. User-Agent: FreeSWITCH. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 201. X-FS-Support: update_display,send_info. Remote-Party-ID: "00000000" <sip:00000000@2.2.2.2>;party=calling;screen=no;privacy=full. . v=0. o=FreeSWITCH 1334887230 1334887231 IN IP4 2.2.2.2. s=FreeSWITCH. c=IN IP4 2.2.2.2. t=0 0. m=audio 21466 RTP/AVP 8 101 13. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2012/04/20 08:58:16.176573 3.3.3.3:5060 -> 2.2.2.2:5080 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS. From: "00000000" <sip:00000000@2.2.2.2>;tag=4vjXj69rtpaeS. To: <sip:1234@3.3.3.3:5060>. Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b. CSeq: 27118284 INVITE. User-Agent: SBC01. Content-Length: 0. . U 2012/04/20 08:58:16.282935 3.3.3.3:5060 -> 2.2.2.2:5080 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS. From: "00000000" <sip:00000000@2.2.2.2>;tag=4vjXj69rtpaeS. To: <sip:1234@3.3.3.3:5060>;tag=7NNjeSyyjtHSB. Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b. CSeq: 27118284 INVITE. Contact: <sip:05600660128@3.3.3.3:5060;transport=udp>. User-Agent: SBC01. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0.
Answered my own question... the problem occurrs when using call recording on a channel. With the dial plan command: <action application="record_session" data="somefile.wav"/> the problem is experienced. If we set call recording to only happen on the B-Leg using bridge: <action application="bridge" data="{execute_on_answer=record_session 'somefile.wav'}sofia/gateway/gateway/1234"/> then the problem doesnt exist. I guess this is because FS is settling on a codec in order to start call recording, so the only way around this would be for FS to generate the ringing. Im sure there is a way to do this, but in my particular setup this is not required, and setting the recording on the b-leg works just fine. Many thanks On 20 Apr 2012, at 09:14, Barnaby Ritchley wrote:
Hi Guys
I have a situation where a call is received on a-leg, FS sends 183 session progress to the a-leg before the call is attempted in b-leg.
Then, when the b-leg attempts the call for the bridge and sends an invite, it receives 180 ringing, and this is not relayed back through the a leg, so the a-leg never hears ringing.
The call goes:
Cisco UA (1.1.1.1) -> FS (2.2.2.2) -> Gateway (3.3.3.3)
Sofia config has nothing special in it, relevent sections:
<param name="inbound-codec-negotiation" value="generous"/> <param name="disable-transcoding" value="true"/> <param name="manual-redirect" value="true"/> <param name="disable-transfer" value="true"/>
and the dial plan is straight forward, just a bridge between the two endpoints.
Any ideas?
Here is the trace of the call:
INVITE sip:1234@2.2.2.2:5060 SIP/2.0. Date: Fri, 20 Apr 2012 07:58:36 GMT. Call-Info: <sip:1.1.1.1:5060>;method="NOTIFY;Event=telephone-event;Duration=500". Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY. From: "User" <sip:00000000@1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567. Allow-Events: presence, kpml. Supported: 100rel,timer,resource-priority,replaces. Supported: X-cisco-srtp-fallback. Supported: Geolocation. Min-SE: 1800. Content-Length: 212. To: <sip:1234@2.2.2.2>. Expires: 180. Content-Type: application/sdp. Call-ID: a088a380-f911172c-3d33e6-8d200795@1.1.1.1. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979. CSeq: 101 INVITE. Session-Expires: 1800. Max-Forwards: 69. . v=0. o=CiscoSystemsCCM-SIP 2000 1 IN IP4 1.1.1.1. s=SIP Call. c=IN IP4 1.1.1.1. t=0 0. m=audio 31422 RTP/AVP 8 101. a=rtpmap:8 PCMA/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15.
U 2012/04/20 08:58:16.142711 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979. From: "User" <sip:00000000@1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567. To: <sip:1234@2.2.2.2>. CSeq: 101 INVITE. User-Agent: FreeSWITCH. Content-Length: 0. .
U 2012/04/20 08:58:16.169453 2.2.2.2:5060 -> 1.1.1.1:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK7ccc6a3fe8979. From: "User" <sip:00000000@1.1.1.1>;tag=467b6df4-0e4d-47d0-b237-533fec74e710-20598567. To: <sip:1234@2.2.2.2>;tag=vm45yDZjgmDKH. Call-ID: a088a380-f911172c-3d33e6-8d200795@1.1.1.1. CSeq: 101 INVITE. Contact: <sip:1234@2.2.2.2:5060;transport=udp>. User-Agent: FreeSWITCH. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 247. Remote-Party-ID: "1234" <sip:1234@2.2.2.2>;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1334881876 1334881877 IN IP4 2.2.2.2. s=FreeSWITCH. c=IN IP4 2.2.2.2. t=0 0. m=audio 26820 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.
U 2012/04/20 08:58:16.176126 2.2.2.2:5080 -> 3.3.3.3:5060 INVITE sip:05600660128@3.3.3.3:5060 SIP/2.0. Via: SIP/2.0/UDP 2.2.2.2:5080;rport;branch=z9hG4bK44BS1jH9Za6yS. Max-Forwards: 68. From: "00000000" <sip:00000000@2.2.2.2>;tag=4vjXj69rtpaeS. To: <sip:1234@3.3.3.3:5060>. Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b. CSeq: 27118284 INVITE. Contact: <sip:gw+C4DF4B2A088C91@2.2.2.2:5080;transport=udp;gw=C4DF4B2A088C91>. User-Agent: FreeSWITCH. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 201. X-FS-Support: update_display,send_info. Remote-Party-ID: "00000000" <sip:00000000@2.2.2.2>;party=calling;screen=no;privacy=full. . v=0. o=FreeSWITCH 1334887230 1334887231 IN IP4 2.2.2.2. s=FreeSWITCH. c=IN IP4 2.2.2.2. t=0 0. m=audio 21466 RTP/AVP 8 101 13. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20.
U 2012/04/20 08:58:16.176573 3.3.3.3:5060 -> 2.2.2.2:5080 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS. From: "00000000" <sip:00000000@2.2.2.2>;tag=4vjXj69rtpaeS. To: <sip:1234@3.3.3.3:5060>. Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b. CSeq: 27118284 INVITE. User-Agent: SBC01. Content-Length: 0. .
U 2012/04/20 08:58:16.282935 3.3.3.3:5060 -> 2.2.2.2:5080 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 2.2.2.2:5080;rport=5080;branch=z9hG4bK44BS1jH9Za6yS. From: "00000000" <sip:00000000@2.2.2.2>;tag=4vjXj69rtpaeS. To: <sip:1234@3.3.3.3:5060>;tag=7NNjeSyyjtHSB. Call-ID: 6d4df4bc-0561-1230-4e8c-001ec9d8773b. CSeq: 27118284 INVITE. Contact: <sip:05600660128@3.3.3.3:5060;transport=udp>. User-Agent: SBC01. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0.
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participants (1)
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Barnaby Ritchley