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Исчезает ringing из bridge https://freeswitchforum.com/viewtopic.php?f=6&t=1231 |
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Автор: | mic_last [ 16 июн 2022 20:50 ] |
Заголовок сообщения: | Исчезает ringing из bridge |
Здравствуйте все! Помогите пожалуйста решить проблему - при соединении в bridge вызываемая сторона посылает ringing но до вызывающего плеча ringing обратно не доходит. Входящий вызов, в котором куда-то делся ringing: 2022/06/16 12:26:51.977643 192.168.125.44:5060 -> 192.168.240.41:5060 INVITE sip:711111111111116@192.168.240.41 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.44:5060;branch=z9hG4bK1f01f62d2daa1677 From: <sip:+12345678901@192.168.125.44>;tag=6ffe3f921f4a9166 To: <sip:711111111111116@192.168.240.41> Call-ID: 42dcabca663616c6c065b65f4@192.168.125.44 CSeq: 21906 INVITE Contact: <sip:+12345678901@192.168.125.44:5060> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK Supported: timer, linknat Original-Info: EluTIhJeGU8SW5MiUUsBIVFTCUNGVwgGVUEBQwcCVVNaEhcXVV8UQwULVFB/ Max-Forwards: 70 User-Agent: VOS3000 V2.1.9.06 Session-Expires: 14400 Content-Type: application/sdp Content-Length: 218 v=0 o=- 22987 22987 IN IP4 192.168.125.44 s=- c=IN IP4 192.168.125.44 t=0 0 m=audio 36476 RTP/AVP 0 5 101 a=rtpmap:0 PCMU/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 2022/06/16 12:26:51.981185 192.168.240.41:5060 -> 192.168.125.44:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.125.44:5060;branch=z9hG4bK1f01f62d2daa1677 From: <sip:+12345678901@192.168.125.44>;tag=6ffe3f921f4a9166 To: <sip:711111111111116@192.168.240.41> Call-ID: 42dcabca663616c6c065b65f4@192.168.125.44 CSeq: 21906 INVITE User-Agent: FreeSWITCH-mod_sofia/1.10.0~64bit Content-Length: 0 2022/06/16 12:26:53.506903 192.168.240.41:5060 -> 192.168.125.44:5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.125.44:5060;branch=z9hG4bK1f01f62d2daa1677 From: <sip:+12345678901@192.168.125.44>;tag=6ffe3f921f4a9166 To: <sip:711111111111116@192.168.240.41>;tag=60vvcN8Kc0r4a Call-ID: 42dcabca663616c6c065b65f4@192.168.125.44 CSeq: 21906 INVITE Contact: <sip:711111111111116@192.168.240.41:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.10.0~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 218 Remote-Party-ID: "Outbound Call" <sip:111111111116@192.168.240.41>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1655379131 1655379132 IN IP4 192.168.240.41 s=FreeSWITCH c=IN IP4 192.168.240.41 t=0 0 m=audio 17682 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2022/06/16 12:26:56.865191 192.168.240.41:5060 -> 192.168.125.44:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.125.44:5060;branch=z9hG4bK1f01f62d2daa1677 From: <sip:+12345678901@192.168.125.44>;tag=6ffe3f921f4a9166 To: <sip:711111111111116@192.168.240.41>;tag=60vvcN8Kc0r4a Call-ID: 42dcabca663616c6c065b65f4@192.168.125.44 CSeq: 21906 INVITE Contact: <sip:711111111111116@192.168.240.41:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.10.0~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 14400;refresher=uac Content-Type: application/sdp Content-Disposition: session Content-Length: 218 Remote-Party-ID: "Outbound Call" <sip:111111111116@192.168.240.41>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1655379131 1655379132 IN IP4 192.168.240.41 s=FreeSWITCH c=IN IP4 192.168.240.41 t=0 0 m=audio 17682 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2022/06/16 12:26:56.869154 192.168.125.44:5060 -> 192.168.240.41:5060 ACK sip:711111111111116@192.168.240.41:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.125.44:5060;branch=z9hG4bK02a1a2d306809734 From: <sip:+12345678901@192.168.125.44>;tag=6ffe3f921f4a9166 To: <sip:711111111111116@192.168.240.41>;tag=60vvcN8Kc0r4a Call-ID: 42dcabca663616c6c065b65f4@192.168.125.44 CSeq: 21906 ACK Max-Forwards: 70 Content-Length: 0 2022/06/16 12:27:13.506746 192.168.240.41:5060 -> 192.168.125.44:5060 BYE sip:+12345678901@192.168.125.44:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.240.41;rport;branch=z9hG4bKFZUy8DHcDm9ja Max-Forwards: 70 From: <sip:711111111111116@192.168.240.41>;tag=60vvcN8Kc0r4a To: <sip:+12345678901@192.168.125.44>;tag=6ffe3f921f4a9166 Call-ID: 42dcabca663616c6c065b65f4@192.168.125.44 CSeq: 53144624 BYE User-Agent: FreeSWITCH-mod_sofia/1.10.0~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: vos;cause=-8;text="CalleeHangup" Content-Length: 0 2022/06/16 12:27:13.511238 192.168.125.44:5060 -> 192.168.240.41:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.240.41;received=192.168.240.41;rport=5060;branch=z9hG4bKFZUy8DHcDm9ja From: <sip:711111111111116@192.168.240.41>;tag=60vvcN8Kc0r4a To: <sip:+12345678901@192.168.125.44>;tag=6ffe3f921f4a9166 Call-ID: 42dcabca663616c6c065b65f4@192.168.125.44 CSeq: 53144624 BYE Content-Length: 0 Исходящий из бриджа в котором ringing есть: 2022/06/16 12:26:52.308368192.168.230.41:5080 -> 192.168.125.44:5060 INVITE sip:650111111111116@192.168.125.44 SIP/2.0 Via: SIP/2.0/UDP192.168.230.41:5080;rport;branch=z9hG4bKjj25545Z83SFK Max-Forwards: 69 From: "+12345678901" <sip:+12345678901@8.11.230.41>;tag=KUKU5HZU2y2pm To: <sip:650111111111116@192.168.125.44> Call-ID: f878f60a-6833-123b-7d9d-005056a3f7a7 CSeq: 53144614 INVITE Contact: <sip:gw+SS2@8.11.230.41:5080;transport=udp;gw=SS2> User-Agent: FreeSWITCH-mod_sofia/1.10.0~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 266 X-FS-Support: update_display,send_info Remote-Party-ID: "+12345678901" <sip:+12345678901@8.11.230.41>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1655364608 1655364609 IN IP4192.168.230.41 s=FreeSWITCH c=IN IP4192.168.230.41 t=0 0 m=audio 32204 RTP/AVP 8 0 101 13 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 2022/06/16 12:26:52.312488 192.168.125.44:5060 ->192.168.230.41:5080 SIP/2.0 100 Trying Via: SIP/2.0/UDP192.168.230.41:5080;received=8.11.230.41;rport=5080;branch=z9hG4bKjj25545Z83SFK From: "+12345678901" <sip:+12345678901@8.11.230.41>;tag=KUKU5HZU2y2pm To: <sip:650111111111116@192.168.125.44>;tag=5bdc9d991a63d39a Call-ID: f878f60a-6833-123b-7d9d-005056a3f7a7 CSeq: 53144614 INVITE Server: VOS3000 V2.1.9.06 Content-Length: 0 2022/06/16 12:26:53.497475 192.168.125.44:5060 ->192.168.230.41:5080 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP192.168.230.41:5080;received=8.11.230.41;rport=5080;branch=z9hG4bKjj25545Z83SFK From: "+12345678901" <sip:+12345678901@8.11.230.41>;tag=KUKU5HZU2y2pm To: <sip:650111111111116@192.168.125.44>;tag=5bdc9d991a63d39a Call-ID: f878f60a-6833-123b-7d9d-005056a3f7a7 CSeq: 53144614 INVITE Contact: <sip:650111111111116@192.168.125.44:5060> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK Server: VOS3000 V2.1.9.06 Content-Type: application/sdp Content-Length: 206 v=0 o=- 22988 22989 IN IP4 198.241.13.15 s=- c=IN IP4 198.241.13.15 t=0 0 m=audio 30280 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2022/06/16 12:26:53.710860 192.168.125.44:5060 ->192.168.230.41:5080 SIP/2.0 180 Ringing Via: SIP/2.0/UDP192.168.230.41:5080;received=8.11.230.41;rport=5080;branch=z9hG4bKjj25545Z83SFK From: "+12345678901" <sip:+12345678901@8.11.230.41>;tag=KUKU5HZU2y2pm To: <sip:650111111111116@192.168.125.44>;tag=5bdc9d991a63d39a Call-ID: f878f60a-6833-123b-7d9d-005056a3f7a7 CSeq: 53144614 INVITE Contact: <sip:650111111111116@192.168.125.44:5060> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK Server: VOS3000 V2.1.9.06 Content-Type: application/sdp Content-Length: 206 v=0 o=- 22988 22989 IN IP4 198.241.13.15 s=- c=IN IP4 198.241.13.15 t=0 0 m=audio 30280 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2022/06/16 12:26:56.819977 192.168.125.44:5060 ->192.168.230.41:5080 SIP/2.0 200 OK Via: SIP/2.0/UDP192.168.230.41:5080;received=8.11.230.41;rport=5080;branch=z9hG4bKjj25545Z83SFK From: "+12345678901" <sip:+12345678901@8.11.230.41>;tag=KUKU5HZU2y2pm To: <sip:650111111111116@192.168.125.44>;tag=5bdc9d991a63d39a Call-ID: f878f60a-6833-123b-7d9d-005056a3f7a7 CSeq: 53144614 INVITE Contact: <sip:650111111111116@192.168.125.44:5060> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, UPDATE, PRACK Server: VOS3000 V2.1.9.06 Supported: timer Session-Expires: 14400;refresher=uas Content-Type: application/sdp Content-Length: 206 v=0 o=- 22988 22989 IN IP4 198.241.13.15 s=- c=IN IP4 198.241.13.15 t=0 0 m=audio 30280 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2022/06/16 12:26:56.841592192.168.230.41:5080 -> 192.168.125.44:5060 ACK sip:650111111111116@192.168.125.44:5060 SIP/2.0 Via: SIP/2.0/UDP192.168.230.41:5080;rport;branch=z9hG4bKXNrpSeZ685eXj Max-Forwards: 70 From: "+12345678901" <sip:+12345678901@8.11.230.41>;tag=KUKU5HZU2y2pm To: <sip:650111111111116@192.168.125.44>;tag=5bdc9d991a63d39a Call-ID: f878f60a-6833-123b-7d9d-005056a3f7a7 CSeq: 53144614 ACK Contact: <sip:gw+SS2@8.11.230.41:5080;transport=udp;gw=SS2> Content-Length: 0 2022/06/16 12:27:13.498049 192.168.125.44:5060 ->192.168.230.41:5080 BYE sip:gw+SS2@8.11.230.41:5080;transport=udp;gw=SS2 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.44:5060;branch=z9hG4bK4736b1c57bdebb8d From: <sip:650111111111116@192.168.125.44>;tag=5bdc9d991a63d39a To: "+12345678901" <sip:+12345678901@8.11.230.41>;tag=KUKU5HZU2y2pm Call-ID: f878f60a-6833-123b-7d9d-005056a3f7a7 CSeq: 21906 BYE Max-Forwards: 70 Reason: vos;cause=-8;text="CalleeHangup" Content-Length: 0 2022/06/16 12:27:13.501460192.168.230.41:5080 -> 192.168.125.44:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.125.44:5060;branch=z9hG4bK4736b1c57bdebb8d From: <sip:650111111111116@192.168.125.44>;tag=5bdc9d991a63d39a To: "+12345678901" <sip:+12345678901@8.11.230.41>;tag=KUKU5HZU2y2pm Call-ID: f878f60a-6833-123b-7d9d-005056a3f7a7 CSeq: 21906 BYE User-Agent: FreeSWITCH-mod_sofia/1.10.0~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Length: 0 |
Автор: | Vlad1983 [ 20 июн 2022 09:57 ] |
Заголовок сообщения: | Re: Исчезает ringing из bridge |
ждете SIP/2.0 180 Ringing в вызывающем плече? от вызываемого прилетает SIP/2.0 183 Session Progress c SDP после него не важно есть SIP/2.0 180 Ringing или нет медиа уже проключается с вызываемой стороны попробовать выставить в дридж "{ignore_early_media=consume}" |
Автор: | mic_last [ 20 июн 2022 10:41 ] |
Заголовок сообщения: | Re: Исчезает ringing из bridge |
Привет! У меня нет проблемы с медиа - моя задача сделать прозрачное транскодирование, а именно: все что от вызываемой стороны прилетает то и передать вызывающей стороне, но только с другими кодеками. Что удивительно Asterisk это делает а как добиться правды от FreeSWITCH я пока не нашел... Диалплан выглядит так: <extension name="public_711"> <condition field="destination_number" expression="^711(\d+)$"> <action application="set" data="domain_name=$${domain}"/> <action application="set" data="outbound_caller_id_number=${caller_id_number}"/> <action application="log" data="WARNING CID Num is ${outbound_caller_id_number}"/> <action application="set" data="absolute_codec_string=PCMA,PCMU"/> <action application="bridge" data="{absolute_codec_string='PCMA,PCMU'}sofia/gateway/SS2/$1"/> </condition> </extension> Vlad1983 писал(а): ждете SIP/2.0 180 Ringing в вызывающем плече?
от вызываемого прилетает SIP/2.0 183 Session Progress c SDP после него не важно есть SIP/2.0 180 Ringing или нет медиа уже проключается с вызываемой стороны попробовать выставить в дридж "{ignore_early_media=consume}" |
Автор: | Vlad1983 [ 20 июн 2022 15:50 ] |
Заголовок сообщения: | Re: Исчезает ringing из bridge |
человек наоборот жалуется что у него транскодит Early Media and "Late Negotiation" |
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