- FreeSWITCH-users - FreeSWITCH Mailing Lists

Wiki editing account
by David McNab 25 Aug '15

25 Aug '15

Hi guys, How do I go about applying for an editing account on the wiki? In my learning process, I've been figuring out some things about Python scripting which might be able to help others who are Python-inclined. I'm asking if it might be possible to get edit access on the wiki, so I can publish some walk-through examples, together with practical tips and caveats. Cheers David

2 1

Need help to convert RTP/SAVP to RTP/AVP
by Pete Kay 25 Aug '15

25 Aug '15

Hi I am trying to send call from webrtc to freeswitch and then bridge to an external endpoint that is a normal SIP phone. I tried to use sip_secure_media=false to force the media to be converted from SAVP to AVP. But the outgoing INVITE is still showing SAVP ( sip trace blow ). Does anyone know where I am doing wrong? any help will be greately appreciated. 2015-08-17 15:29:46.341244 [INFO] mod_dialplan_xml.c:637 Processing 1001 <1001>->1002 in context default Dialplan: sofia/internal/1001@y.y.y.y:5060 parsing [default->incoming_calls] continue=false Dialplan: sofia/internal/1001@y.y.y.y:5060 Regex (PASS) [incoming_calls] ${destination_number}(1002) =~ /1002/ break=on-false Dialplan: sofia/internal/1001@y.y.y.y:5060 Action bridge({sip_secure_media=false}[sip_secure_media=false]sofia/internal/754617162742492(a)x.x.x.x ) 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:216 (sofia/internal/1001@y.y.y.y:5060) State Change CS_ROUTING -> CS_EXECUTE 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/1001@y.y.y.y:5060) State ROUTING going to sleep 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001@y.y.y.y:5060) Running State Change CS_EXECUTE 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1001@y.y.y.y:5060) State EXECUTE 2015-08-17 15:29:46.341244 [DEBUG] mod_sofia.c:196 sofia/internal/1001@y.y.y.y:5060 SOFIA EXECUTE 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:258 sofia/internal/1001@y.y.y.y:5060 Standard EXECUTE EXECUTE sofia/internal/1001@y.y.y.y:5060 bridge({sip_secure_media=false}[sip_secure_media=false]sofia/internal/754617162742492(a)x.x.x.x ) 2015-08-17 15:29:46.341244 [DEBUG] switch_ivr_originate.c:2104 Parsing global variables 2015-08-17 15:29:46.341244 [DEBUG] switch_ivr_originate.c:2557 Parsing session specific variables 2015-08-17 15:29:46.341244 [NOTICE] switch_channel.c:1089 New Channel sofia/internal/754617162742492(a)x.x.x.x [ca6d5930-44f4-11e5-9cc9-c9574f94e170] 2015-08-17 15:29:46.341244 [DEBUG] mod_sofia.c:4735 (sofia/internal/754617162742492(a)x.x.x.x) State Change CS_NEW -> CS_INIT 2015-08-17 15:29:46.341244 [DEBUG] mod_sofia.c:4805 [zrtp_passthru] Setting a-leg inherit_codec=true 2015-08-17 15:29:46.341244 [DEBUG] mod_sofia.c:4808 [zrtp_passthru] Setting b-leg absolute_codec_string='mod_spandsp.G722@8000h@20i@64000b ,CORE_PCM_MODULE.PCMU@8000h@20i@64000b,CORE_PCM_MODULE.PCMA@8000h@20i@64000b ' 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/754617162742492(a)x.x.x.x) Running State Change CS_INIT 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/754617162742492(a)x.x.x.x) State INIT 2015-08-17 15:29:46.341244 [DEBUG] mod_sofia.c:88 sofia/internal/754617162742492(a)x.x.x.x SOFIA INIT 2015-08-17 15:29:46.341244 [DEBUG] sofia_glue.c:1257 sofia/internal/754617162742492(a)x.x.x.x sending invite version: 1.7.0 git 58f1272 2015-08-14 17:51:12Z 64bit Local SDP: v=0 o=- 4133111956976529630 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS media_stream m=audio 9 RTP/SAVPF 111 103 9 102 0 8 106 105 13 127 126 c=IN IP4 0.0.0.0 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:9 G722/8000 a=rtpmap:102 ILBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:127 red/8000 a=rtpmap:126 telephone-event/8000 a=rtcp:9 IN IP4 0.0.0.0 a=ice-ufrag:Gn8CX52vaidE1C73 a=ice-pwd:16VDNSZodn+NJp8u3O1mQrMy a=fingerprint:sha-1 19:57:5D:7A:BD:7B:42:3B:C9:1B:2F:A6:D2:4F:32:FD:C3:23:14:D9 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=rtcp-mux a=maxptime:60 a=ssrc:3618235845 cname:3hHDnnOtFofiuRz2 a=ssrc:3618235845 msid:media_stream audio_track a=ssrc:3618235845 mslabel:media_stream a=ssrc:3618235845 label:audio_track a=candidate:186199869 1 udp 2122260223 192.168.1.101 60710 typ host generation 0 a=candidate:4049598204 1 udp 2122194687 10.73.229.110 52079 typ host generation 0 a=candidate:186199869 2 udp 2122260222 192.168.1.101 64150 typ host generation 0 a=candidate:4049598204 2 udp 2122194686 10.73.229.110 51909 typ host generation 0 a=candidate:1167774669 1 tcp 1518280447 192.168.1.101 55638 typ host tcptype passive generation 0 a=candidate:3218868748 1 tcp 1518214911 10.73.229.110 55639 typ host tcptype passive generation 0 a=candidate:1167774669 2 tcp 1518280446 192.168.1.101 55640 typ host tcptype passive generation 0 a=candidate:3218868748 2 tcp 1518214910 10.73.229.110 55641 typ host tcptype passive generation 0 a=candidate:2320574857 1 udp 1686052607 27.38.56.3 36782 typ srflx raddr 192.168.1.101 rport 60710 generation 0 a=candidate:2320574857 2 udp 1686052606 27.38.56.3 37553 typ srflx raddr 192.168.1.101 rport 64150 generation 0 a=candidate:1880617032 1 udp 1685987071 112.97.38.179 2308 typ srflx raddr 10.73.229.110 rport 52079 generation 0 a=candidate:1880617032 2 udp 1685987070 112.97.38.179 2307 typ srflx raddr 10.73.229.110 rport 51909 generation 0 a=candidate:2922827951 1 udp 41885439 146.185.149.11 55532 typ relay raddr 27.38.56.3 rport 36782 generation 0 a=candidate:2922827951 2 udp 41885438 146.185.149.11 54104 typ relay raddr 27.38.56.3 rport 37553 generation 0 a=candidate:2922827951 2 udp 41819902 146.185.149.11 63311 typ relay raddr 112.97.38.179 rport 54091 generation 0 a=candidate:2922827951 1 udp 41819903 146.185.149.11 49576 typ relay raddr 112.97.38.179 rport 54092 generation 0 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:40 sofia/internal/754617162742492(a)x.x.x.x Standard INIT 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/754617162742492(a)x.x.x.x) State Change CS_INIT -> CS_ROUTING 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/754617162742492(a)x.x.x.x) State INIT going to sleep 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/754617162742492(a)x.x.x.x) Running State Change CS_ROUTING 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/754617162742492(a)x.x.x.x) State ROUTING 2015-08-17 15:29:46.341244 [DEBUG] mod_sofia.c:141 sofia/internal/754617162742492(a)x.x.x.x SOFIA ROUTING 2015-08-17 15:29:46.341244 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/754617162742492(a)x.x.x.x) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/754617162742492(a)x.x.x.x) State ROUTING going to sleep 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/754617162742492(a)x.x.x.x) Running State Change CS_CONSUME_MEDIA 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/754617162742492(a)x.x.x.x) State CONSUME_MEDIA 2015-08-17 15:29:46.341244 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/754617162742492(a)x.x.x.x) State CONSUME_MEDIA going to sleep send 3635 bytes to udp/[69.27.168.50]:5060 at 15:29:46.355875: ------------------------------------------------------------------------ INVITE sip:754617162742492@x.x.x.x SIP/2.0 Via: SIP/2.0/UDP y.y.y.y;rport;branch=z9hG4bKcSe6jHj8QS56e Max-Forwards: 69 From: "Extension 1001" <sip:1001@y.y.y.y>;tag=H6HrjU4ZH1KjD To: <sip:754617162742492@x.x.x.x> Call-ID: a1cbcf13-bf97-1233-84b6-60eb698211be CSeq: 79576629 INVITE Contact: <sip:mod_sofia@y.y.y.y:5060> User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20150814T175112Z~58f1272490~64bit 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: 2665 X-FS-Support: update_display,send_info Remote-Party-ID: "Extension 1001" <sip:1001@y.y.y.y >;party=calling;screen=yes;privacy=off v=0 o=- 4133111956976529630 2 IN IP4 y.y.y.y s=- t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS media_stream m=audio 9 RTP/SAVPF 111 103 9 102 0 8 106 105 13 127 126 c=IN IP4 0.0.0.0 a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10; useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:9 G722/8000 a=rtpmap:102 ILBC/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:127 red/8000 a=rtpmap:126 telephone-event/8000 a=rtcp:9 IN IP4 0.0.0.0 a=ice-ufrag:Gn8CX52vaidE1C73 a=ice-pwd:16VDNSZodn+NJp8u3O1mQrMy a=fingerprint:sha-1 19:57:5D:7A:BD:7B:42:3B:C9:1B:2F:A6:D2:4F:32:FD:C3:23:14:D9 a=setup:actpass a=mid:audio a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=rtcp-mux a=maxptime:60 a=ssrc:3618235845 cname:3hHDnnOtFofiuRz2 a=ssrc:3618235845 msid:media_stream audio_track a=ssrc:3618235845 mslabel:media_stream a=ssrc:3618235845 label:audio_track a=candidate:186199869 1 udp 2122260223 192.168.1.101 60710 typ host generation 0 a=candidate:4049598204 1 udp 2122194687 10.73.229.110 52079 typ host generation 0 a=candidate:186199869 2 udp 2122260222 192.168.1.101 64150 typ host generation 0 a=candidate:4049598204 2 udp 2122194686 10.73.229.110 51909 typ host generation 0 a=candidate:1167774669 1 tcp 1518280447 192.168.1.101 55638 typ host tcptype passive generation 0 a=candidate:3218868748 1 tcp 1518214911 10.73.229.110 55639 typ host tcptype passive generation 0 a=candidate:1167774669 2 tcp 1518280446 192.168.1.101 55640 typ host tcptype passive generation 0 a=candidate:3218868748 2 tcp 1518214910 10.73.229.110 55641 typ host tcptype passive generation 0 a=candidate:2320574857 1 udp 1686052607 27.38.56.3 36782 typ srflx raddr 192.168.1.101 rport 60710 generation 0 a=candidate:2320574857 2 udp 1686052606 27.38.56.3 37553 typ srflx raddr 192.168.1.101 rport 64150 generation 0 a=candidate:1880617032 1 udp 1685987071 112.97.38.179 2308 typ srflx raddr 10.73.229.110 rport 52079 generation 0 a=candidate:1880617032 2 udp 1685987070 112.97.38.179 2307 typ srflx raddr 10.73.229.110 rport 51909 generation 0 a=candidate:2922827951 1 udp 41885439 146.185.149.11 55532 typ relay raddr 27.38.56.3 rport 36782 generation 0 a=candidate:2922827951 2 udp 41885438 146.185.149.11 54104 typ relay raddr 27.38.56.3 rport 37553 generation 0 a=candidate:2922827951 2 udp 41819902 146.185.149.11 63311 typ relay raddr 112.97.38.179 rport 54091 generation 0 a=candidate:2922827951 1 udp 41819903 146.185.149.11 49576 typ relay raddr 112.97.38.179 rport 54092 generation 0 ------------------------------------------------------------------------ 2015-08-17 15:29:46.341244 [DEBUG] sofia.c:6715 Channel sofia/internal/754617162742492(a)x.x.x.x entering state [calling][0] recv 308 bytes from udp/[x.x.x.x]:5060 at 15:29:46.357317: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP y.y.y.y;rport;branch=z9hG4bKcSe6jHj8QS56e From: "Extension 1001" <sip:1001@y.y.y.y>;tag=H6HrjU4ZH1KjD To: <sip:754617162742492@x.x.x.x> Call-ID: a1cbcf13-bf97-1233-84b6-60eb698211be CSeq: 79576629 INVITE User-Agent: DNL-Switch Content-Length: 0 ------------------------------------------------------------------------ recv 433 bytes from udp/[x.x.x.x]:5060 at 15:29:46.367995: ------------------------------------------------------------------------ SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP y.y.y.y;rport;branch=z9hG4bKcSe6jHj8QS56e From: "Extension 1001" <sip:1001@y.y.y.y>;tag=H6HrjU4ZH1KjD To: <sip:754617162742492@x.x.x.x>;tag=Ut8ye62NUrQeB Call-ID: a1cbcf13-bf97-1233-84b6-60eb698211be CSeq: 79576629 INVITE User-Agent: DNL-Switch Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO Supported: timer Content-Length: 0 ------------------------------------------------------------------------ send 343 bytes to udp/[x.x.x.x]:5060 at 15:29:46.368134: ------------------------------------------------------------------------ ACK sip:754617162742492@x.x.x.x SIP/2.0 Via: SIP/2.0/UDP y.y.y.y;rport;branch=z9hG4bKcSe6jHj8QS56e Max-Forwards: 69 From: "Extension 1001" <sip:1001@y.y.y.y>;tag=H6HrjU4ZH1KjD To: <sip:754617162742492@x.x.x.x>;tag=Ut8ye62NUrQeB Call-ID: a1cbcf13-bf97-1233-84b6-60eb698211be CSeq: 79576629 ACK Content-Length: 0 ------------------------------------------------------------------------ 2015-08-17 15:29:46.361276 [DEBUG] sofia.c:6715 Channel sofia/internal/754617162742492(a)x.x.x.x entering state [terminated][415] 2015-08-17 15:29:46.361276 [NOTICE] sofia.c:7631 Hangup sofia/internal/754617162742492(a)x.x.x.x [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/754617162742492(a)x.x.x.x) Running State Change CS_HANGUP 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/754617162742492(a)x.x.x.x) Callstate Change DOWN -> HANGUP 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/754617162742492(a)x.x.x.x) State HANGUP 2015-08-17 15:29:46.361276 [DEBUG] mod_sofia.c:431 Channel sofia/internal/754617162742492(a)x.x.x.x hanging up, cause: SERVICE_NOT_IMPLEMENTED 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:60 sofia/internal/754617162742492(a)x.x.x.x Standard HANGUP, cause: SERVICE_NOT_IMPLEMENTED 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/754617162742492(a)x.x.x.x) State HANGUP going to sleep 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/754617162742492(a)x.x.x.x) State Change CS_HANGUP -> CS_REPORTING 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/754617162742492(a)x.x.x.x) Running State Change CS_REPORTING 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/754617162742492(a)x.x.x.x) State REPORTING 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:104 sofia/internal/754617162742492(a)x.x.x.x Standard REPORTING, cause: SERVICE_NOT_IMPLEMENTED 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/754617162742492(a)x.x.x.x) State REPORTING going to sleep 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/754617162742492(a)x.x.x.x) State Change CS_REPORTING -> CS_DESTROY 2015-08-17 15:29:46.361276 [DEBUG] switch_core_session.c:1639 Session 2 (sofia/internal/754617162742492(a)x.x.x.x) Locked, Waiting on external entities 2015-08-17 15:29:46.361276 [DEBUG] switch_ivr_originate.c:3727 Originate Resulted in Error Cause: 79 [SERVICE_NOT_IMPLEMENTED] 2015-08-17 15:29:46.361276 [NOTICE] switch_core_session.c:1657 Session 2 (sofia/internal/754617162742492(a)x.x.x.x) Ended 2015-08-17 15:29:46.361276 [NOTICE] switch_core_session.c:1661 Close Channel sofia/internal/754617162742492(a)x.x.x.x [CS_DESTROY] 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/754617162742492(a)x.x.x.x) Running State Change CS_DESTROY 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/754617162742492(a)x.x.x.x) State DESTROY 2015-08-17 15:29:46.361276 [DEBUG] mod_sofia.c:341 sofia/internal/754617162742492(a)x.x.x.x SOFIA DESTROY 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:111 sofia/internal/754617162742492(a)x.x.x.x Standard DESTROY 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/754617162742492(a)x.x.x.x) State DESTROY going to sleep 2015-08-17 15:29:46.361276 [INFO] mod_dptools.c:3333 Originate Failed. Cause: SERVICE_NOT_IMPLEMENTED 2015-08-17 15:29:46.361276 [NOTICE] switch_channel.c:4800 Hangup sofia/internal/1001@y.y.y.y:5060 [CS_EXECUTE] [SERVICE_NOT_IMPLEMENTED] 2015-08-17 15:29:46.361276 [DEBUG] switch_core_session.c:2926 sofia/internal/1001@y.y.y.y:5060 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:539 (sofia/internal/1001@y.y.y.y:5060) State EXECUTE going to sleep 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001@y.y.y.y:5060) Running State Change CS_HANGUP 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/1001@y.y.y.y:5060) Callstate Change RINGING -> HANGUP 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1001@y.y.y.y:5060) State HANGUP 2015-08-17 15:29:46.361276 [DEBUG] mod_sofia.c:425 sofia/internal/1001@y.y.y.y:5060 Overriding SIP cause 501 with 415 from the other leg 2015-08-17 15:29:46.361276 [DEBUG] mod_sofia.c:431 Channel sofia/internal/1001@y.y.y.y:5060 hanging up, cause: SERVICE_NOT_IMPLEMENTED 2015-08-17 15:29:46.361276 [DEBUG] mod_sofia.c:568 Responding to INVITE with: 415 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1001@y.y.y.y:5060 Standard HANGUP, cause: SERVICE_NOT_IMPLEMENTED 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/1001@y.y.y.y:5060) State HANGUP going to sleep 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/1001@y.y.y.y:5060) State Change CS_HANGUP -> CS_REPORTING 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/1001@y.y.y.y:5060) Running State Change CS_REPORTING 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1001@y.y.y.y:5060) State REPORTING send 900 bytes to tcp/[27.38.56.3]:36982 at 15:29:46.376821: ------------------------------------------------------------------------ SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/TCP 192.168.1.101:5060 ;branch=z9hG4bK-524287-1---96ba2f7825316328;rport=36982;received=27.38.56.3 Max-Forwards: 70 From: <sip:1001@y.y.y.y:5060>;tag=9de53b6c To: <sip:1002@y.y.y.y:5060>;tag=gXrZg0KvmrXZH Call-ID: kVdW6FArCGqcEPe7zR3x4g.. CSeq: 2 INVITE User-Agent: FreeSWITCH-mod_sofia/1.7.0+git~20150814T175112Z~58f1272490~64bit 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 Reason: Q.850;cause=79;text="SERVICE_NOT_IMPLEMENTED" Content-Length: 0 Remote-Party-ID: "1002" <sip:1002@y.y.y.y >;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1001@y.y.y.y:5060 Standard REPORTING, cause: SERVICE_NOT_IMPLEMENTED 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/1001@y.y.y.y:5060) State REPORTING going to sleep 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/1001@y.y.y.y:5060) State Change CS_REPORTING -> CS_DESTROY 2015-08-17 15:29:46.361276 [DEBUG] switch_core_session.c:1639 Session 1 (sofia/internal/1001@y.y.y.y:5060) Locked, Waiting on external entities 2015-08-17 15:29:46.361276 [NOTICE] switch_core_session.c:1657 Session 1 (sofia/internal/1001@y.y.y.y:5060) Ended 2015-08-17 15:29:46.361276 [NOTICE] switch_core_session.c:1661 Close Channel sofia/internal/1001@y.y.y.y:5060 [CS_DESTROY] 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/1001@y.y.y.y:5060) Running State Change CS_DESTROY 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1001@y.y.y.y:5060) State DESTROY 2015-08-17 15:29:46.361276 [DEBUG] mod_sofia.c:341 sofia/internal/1001@y.y.y.y:5060 SOFIA DESTROY 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:111 sofia/internal/1001@y.y.y.y:5060 Standard DESTROY 2015-08-17 15:29:46.361276 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/1001@y.y.y.y:5060) State DESTROY going to sleep recv 317 bytes from tcp/[27.38.56.3]:36982 at 15:29:46.651249: ------------------------------------------------------------------------ ACK sip:1002@y.y.y.y:5060 SIP/2.0 Via: SIP/2.0/TCP 192.168.1.101:5060 ;branch=z9hG4bK-524287-1---96ba2f7825316328;rport Max-Forwards: 70 To: <sip:1002@y.y.y.y:5060>;tag=gXrZg0KvmrXZH From: <sip:1001@y.y.y.y:5060>;tag=9de53b6c Call-ID: kVdW6FArCGqcEPe7zR3x4g.. CSeq: 2 ACK Content-Length: 0 ------------------------------------------------------------------------

6 21

Mysql HEX
by Miha 25 Aug '15

25 Aug '15

Hi, I would need a little help with UNHEX thing for file query.conf which is for mysql. Here is one part that is bothering me from query.conf. I posted it on pastebin: http://pastebin.com/JXwuc3ST In accounting request I get like this: CALLEDSTATIONID_WITH_PREFIX = '88#' but in mysql insert radius is inserting like this '88=23' value ''88#' which is HEX. OK how to write this part so that I will be inserted like '88#'? '%{CALLEDSTATIONID_WITH_PREFIX}' tnx miha

1 1

How to maintain redundancy in freeswitch servers
by Shisheer Teli 25 Aug '15

25 Aug '15

I have two FreeSWITCH servers. if one server fails how the calls will direct through the second server? -- Regards, Shisheer T

3 2

FreeSWITCH Week in Review (Master Branch) August 15th-August 21st
by Ken Rice 25 Aug '15

25 Aug '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1PvMxaZ FreeSWITCH Week in Review (Master Branch) August 15th-August 21st Hello, again. This past week in the FreeSWITCH master branch we had 56 commits. This week the features are: a wonderful perl script, filebug.pl, to help file bugs from the command line, and more work on improving verto communicator including integration of Gravatars, some filter options, and grunt. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7988 Add a perl script to help file bugs from the command line and add fixbug.pl to tree FS-8009 [verto communicator] Create a grunt project with livereload support. Documentation can be found here. FS-8010 [verto communicator] Add options for googAutoGainControl, googNoiseSuppression, and googHighpassFilter FS-7855 [verto communicator] Pass userVariables back to the live array to allow for displaying the Gravatar associated with a member’s email address Improvements in build system, cross platform support, and packaging: FS-7965 [mod_conference] Fixed an error thrown when compiling with GCC FS-7985 [mod_voicemail] Fixed a compilation error on 32-bit PCC platform FS-8015 [mod_conference] Add project dir to include for mod_conference so it picks up mod_conference.h for Windows The following bugs were squashed: FS-7970 Fixed crash in video_bug_thread caused by double free FS-7971 [mod_opus] Fixed a rate mismatch and correctly advertise telephone-event and CN rates based on the advertised rates of offered codecs FS-7960 Fixed check_ice routine in switch_core_media.c to not use dropped candidates FS-7975 [mod_voicemail] Fix record-greeting event missing VM-Greeting-Path FS-7969 Fixed a segfault due to pthread_setschedparam() on a thread that has exited FS-7962 Fixed sporadic invite/replaces failure FS-8004 Send keyframe on receiving nack with multiple consecutive packets FS-8005 [mod_opus] Fix for rare decoder error when doing PLC, OPUS_GET_LAST_PACKET_DURATION might return 0 FS-8006 Changed the typedef of switch_core_video_thread_callback_func_t for consistency FS-7932 [mod_verto] Removed the param from the getMute function in verto class, not needed on underlying method FS-8008 [mod_verto] Separate verto default config to have sep v4 and v6 listeners FS-8016 [mod_conference] Reduce buffering of video in conference mux FS-7977 [verto communicator] Fixing default resolution and cleaning code FS-7992 [verto communicator] Fixed device list at settings FS-8017 [verto communicator] Fixed uses of serialized verto in local storage FS-7986 [verto communicator] Fix for devices not refreshing if system config changes FS-7998 [verto communicator] Don’t prompt when recovering call, just do it. FS-8003 [verto communicator] Use audioInDevices instead of audioDevices to match verto plugin

1 0

freeswitch memory leakage.
by Aqs Younas 24 Aug '15

24 Aug '15

Hi, users We are using 6 freeswitch instances, 5 freeswitch instances for playing streams with mod_vlc. After 3 to 4 for days, we see these (5) freeswitch taking more than 3gb of momory and even though calls are not more than 25. Usually, some calls stay for more than 2 to 3 hours on some freeswitch. Everytime we have to restart the freeswitch to release the captured memory. But is believed, freeswitch must release the momory when there are no calls. But freeswitch still keeps captured memory. I have attactted the top command result along with this email. Willing to perform any test if it helps tackle the problem. Thanks for your help.

6 13

Change ptime value
by bhavik patel 24 Aug '15

24 Aug '15

Hello Everyone, I want to change ptime configuration. Will anyone assist me to change ptime value 20 to 40 for codec PCMU,PCMA ? FLow : *SIP-PHONE---Ptime(20)-->FS server (Need to change Ptime to 40) ---------->To Provider* NOTE : If I send 40 Ptime From SIP-PHONE then Call quality is Good but for this i need to change all phones' ptime and I don't want to change it manually.So I need to change this From FS Side. Currently I can see below log in freeswitch logs : m=audio 8000 RTP/AVP 98 97 8 0 3 101 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Dialplan: sofia/default/4602534812(a)192.168.1.26 Action bridge(sofia/gateway/localgateway/001***) m=audio 24716 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv I tried different solutions for this issue but no luck. 1. Add below line before Bridge Channel set(absolute_codec_string=PCMU@8000@40i,PCMA@8000@40i) 2. Add below line before Bridge Channel set(absolute_codec_string=PCMU at 8000@40i,PCMA at 8000@40i) 3.Change below parameter in SIP profile 1.outbound-codec-prefs=PCMU,PCMA to outbound-codec-prefs=PCMU@8000 @@40i,PCMU@16000@@40i,PCMU@32000@@40i,PCMA@8000@40i,PCMA@16000 @40i,PCMA@32000@40i 2.outbound-codec-prefs=PCMU,PCMA,rtp-autofix-timing=false 3.outbound-codec-prefs=PCMU,PCMA,absolute_codec_string=PCMU at 40i,PCMA at 40i,rtp-autofix-timing=false 4.outbound-codec-prefs=PCMU at40i,PCMA at 40i 5.outbound-codec-prefs=PCMU at 8000h@40i,PCMA at 8000h@40i ,inbound-late-negotiation=false 6.outbound-codec-prefs=G729,PCMU,PCMA,inbound-late-negotiation=false Please point me where i am doing wrong. -- Thanks, Bhavik Patel

5 7

Docker/Freeswitch/FusionPBX/snom320 no media on outgoing calls
by Jörn Ott 22 Aug '15

22 Aug '15

Hello all, Our company currently uses an asterisk (without GUI) running on Ubuntu 14.04 in a docker environment. As we had a couple of feature requests, I decided to set up Freeswitch/FusionPBX in parallel and later switch over to it. Almost everything works fine despite Docker giving us NAT between the physical network and the container, using different ports as both containers run on the same physical machine and the firewall NATting the physical network to the outside world. I can make calls to asterisk, receive calls from asterisk and receive calls from the outside which go first to asterisk and then are forwarded to freeswitch. The only thing, I can't get to work is outgoing calls from our snom phones to the external SIP provider via freeswitch. Signalling works, but I don't have any media. I tried direct connections between freeswitch and the SIP provider as well as going through asterisk without any success. On asterisk, we have canreinvite=no in the sip connections to the phones and the provider. I tried the following settings on the internal and external SIP trunk to handle the RTP stream through the PBX. I have set the following options: In vars.xml: <X-PRE-PROCESS cmd="set" data="external_rtp_ip=PUBLIC_IP"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=PUBLIC_IP"/> The public IP is natted to the physical hosts IP on the firewall. In Hello all, Our company currently uses an asterisk (without GUI) running on Ubuntu 14.04 in a docker environment. As we had a couple of feature requests, I decided to set up Freeswitch/FusionPBX in parallel and later switch over to it. Almost everything works fine despite Docker giving us NAT between the physical network and the container, using different ports as both containers run on the same physical machine and the firewall NATting the physical network to the outside world. I can make calls to asterisk, receive calls from asterisk and receive calls from the outside which go first to asterisk and then are forwarded to freeswitch. The only thing, I can't get to work is outgoing calls from our snom phones to the external SIP provider via freeswitch. Signalling works, but I don't have any media. I tried direct connections between freeswitch and the SIP provider as well as going through asterisk without any success. I am now running out of ideas on what else I can try to get outgoing calls to work.On asterisk, we have canreinvite=no in the sip connections to the phones and the provider. I tried the following settings on the internal and external SIP trunk to handle the RTP stream through the PBX. I have set the following options: In vars.xml: <X-PRE-PROCESS cmd="set" data="external_rtp_ip=PUBLIC_IP"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=PUBLIC_IP"/> The public IP is natted to the physical hosts IP on the firewall. In sip_profiles/external.xml, I have the following: <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${external_sip_port}"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="zrtp-passthru" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="tls" value="$${external_ssl_enable}"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${external_tls_port}"/> <param name="tls-cert-dir" value="$${external_ssl_dir}"/> <param name="tls-passphrase" value=""/> <param name="tls-verify-date" value="true"/> <param name="tls-verify-policy" value="none"/> <param name="tls-verify-depth" value="2"/> <param name="tls-verify-in-subjects" value=""/> <param name="tls-version" value="$${sip_tls_version}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="dtmf-type" value="rfc2833"/> <param name="bypass-media" value="false"/> <param name="bypass_media_after_bridge" value="false"/> <param name="bypass_media" value="false"/> <param name="bypass-media-after-bridge" value="false"/> <param name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <param name="sip-sticky-contact" value="true"/> <param name="sip_sticky_contact" value="true"/> <param name="sip_ignore_reinvites" value="true"/> I have added bypass-media and bypass-media-after-bridge with underscore and with dash as I found contradicting information on the web. In sip_profiles/internal.xml, I have <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="watchdog-enabled" value="no"/> <param name="watchdog-step-timeout" value="30000"/> <param name="watchdog-event-timeout" value="30000"/> <param name="log-auth-failures" value="true"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${internal_sip_port}"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="hold-music" value="$${hold_music}"/> <param name="apply-nat-acl" value="nat.auto"/> <param name="apply-inbound-acl" value="domains"/> <param name="local-network-acl" value="localnet.auto"/> <param name="record-path" value="$${recordings_dir}"/> <param name="record-template" value="${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.wav"/> <param name="manage-presence" value="true"/> <param name="presence-probe-on-register" value="true"/> <param name="manage-shared-appearance" value="true"/> <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/> <param name="presence-privacy" value="$${presence_privacy}"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="$${internal_ssl_enable}"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${internal_tls_port}"/> <param name="tls-cert-dir" value="$${internal_ssl_dir}"/> <param name="tls-passphrase" value=""/> <param name="context" value="default"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="tls-verify-date" value="true"/> <param name="tls-verify-policy" value="none"/> <param name="tls-verify-depth" value="2"/> <param name="tls-verify-in-subjects" value=""/> <param name="tls-version" value="$${sip_tls_version}"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="$${internal_auth_calls}"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="challenge-realm" value="auto_to"/> <param name="bypass_media" value="false"/> <param name="bypass-media" value="false"/> <param name="bypass_media_after_bridge" value="false"/> <param name="bypass-media-after-bridge" value="false"/> <param name="sip_sticky_contact" value="true"/> <param name="sip-force-contact" value="NDLB-connectile-dysfunction"/> <param name="sip_ignore_reinvites" value="true"/> <param name="ext-rtp-ip" value="PHYSICAL_IP"/> <param name="ext-sip-ip" value="PHYSICAL_IP"/> PHYSICAL_IP is the IP of the physical host on which the docker container runs. The gateway definition looks like this: <gateway name="781c8f7f-5d96-4e9e-8939-423b883cb675"> <param name="password" value="xxxxx"/> <param name="proxy" value="PROVIDER_HOSTNAME"/> <param name="expire-seconds" value="800"/> <param name="register" value="false"/> <param name="retry-seconds" value="30"/> <param name="context" value="public"/> </gateway> Kind regards Joern

2 1

Crypto Problems with NDLB-allow-crypto-in-avp
by Nicola von Thadden 22 Aug '15

22 Aug '15

Hi, I have a connection to a provider which sadly requires NDLB-allow-crypto-in-avp=true to funktion. The setup is like this: FS -- HG -- POTS | GS FS is my freeswitch server (FreeSWITCH Version 1.4.19-10-1~64bit (-10-1 64bit) on debian, from the repos). HG is the SIP server from eventphone (some might know them from the CCC events in Germany). They provide inbound DDI. I also have a phone GS (Grandstream GXP2160) registered to HG. Freeswitch is set to bridge incoming calls to my number to 3200. When calling from GS to FS, everything works. HG sends an RTP/SAVP in the INVITE and the call is working fine: INVITE sip:gw+eventphone-5849@hg:5080;transport=udp;gw=eventphone-5849 SIP/2.0 Max-Forwards: 19 Via: SIP/2.0/UDP 92.222.104.42:5060;rport;branch=z9hG4bK2140280193 From: "nicoduck" <sip:3008@fs>;tag=218198923 To: <sip:gw+eventphone-5849@fs:5080;transport=udp;gw=eventphone-5849> Call-ID: 1962077623(a)92.222.104.42 CSeq: 4570 INVITE User-Agent: EVENTPHONE PBX TrollEdition v1337 - Problem? Contact: <sip:3008@hg:5060> Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, REGISTER, REFER, OPTIONS, INFO Privacy: none Content-Type: application/sdp Content-Length: 317 v=0 o=yate 1434143858 1434143858 IN IP4 hg s=SIP Call c=IN IP4 hg t=0 0 m=audio 18910 RTP/SAVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:917UNiViuv6i4TB08AS1XI6nzAH9OPKQupyne0kK a=encryption:optional And freeswitch starts the call with: (The 100 and 183 are stripped here, I don't think that there is something important in them) SIP/2.0 200 OK Via: SIP/2.0/UDP 92.222.104.42:5060;rport=5060;branch=z9hG4bK2140280193 From: "nicoduck" <sip:3008@hg>;tag=218198923 To: <sip:gw+eventphone-5849@fs:5080;transport=udp;gw=eventphone-5849>;tag=81UHaFemF50FN Call-ID: 1962077623@hg CSeq: 4570 INVITE Contact: <sip:gw+eventphone-5849@fs:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.19-10-1~64bit 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: 307 Remote-Party-ID: "5849" <sip:5849@fs>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1434111633 1434111634 IN IP4 fs s=FreeSWITCH c=IN IP4 fs t=0 0 m=audio 32224 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:YyL9+W/F2EHV9BD5afj3o6T61X7Z116BSIT+Zlzd FS initiates the crypto and also provides a port for the audio. Audio is working in both directions. The problem starts when I get an incoming POTS call trhough HG. I don't know why but their yate sends a=crypto in a RTP/AVP package. I tried to discuss that problem with the yate developers but they did not seem to be interested in that at all and also don't think that they do something wrong. I added NDLB-allow-crypto-in-avp to my public profile to make freeswitch accept the call. But freeswitch somewhow messes up the crypto when trying to establish a connection: INVITE sip:gw+eventphone-5849@fs:5080;transport=udp;gw=eventphone-5849 SIP/2.0 Max-Forwards: 19 Via: SIP/2.0/UDP hg:5060;rport;branch=z9hG4bK715484803 From: "004917xxxx" <sip:004917xxxx@hg>;tag=776909967 To: <sip:gw+eventphone-5849@fs:5080;transport=udp;gw=eventphone-5849> Call-ID: 555992785@hg CSeq: 4550 INVITE User-Agent: EVENTPHONE PBX TrollEdition v1337 - Problem? Contact: <sip:004917xxxx@hg:5060> Allow: ACK, INVITE, BYE, CANCEL, MESSAGE, REGISTER, REFER, OPTIONS, INFO Content-Type: application/sdp Content-Length: 316 v=0 o=yate 1434139910 1434139910 IN IP4 hg s=SIP Call c=IN IP4 hg t=0 0 m=audio 17678 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:KyzcjFG0m80I8CjQzjQGQCTctNyJo7NZgAyBPoW8 a=encryption:optional The Invite looks okay, besides the a=encryption in the RTP/AVP. This is the answer from the freeswitch: SIP/2.0 200 OK Via: SIP/2.0/UDP hg:5060;rport=5060;branch=z9hG4bK715484803 From: "004917xxxx" <sip:004917xxxx@hg>;tag=776909967 To: <sip:gw+eventphone-5849@fs:5080;transport=udp;gw=eventphone-5849>;tag=FgQ1HF20BNrQp Call-ID: 555992785@hg CSeq: 4550 INVITE Contact: <sip:gw+eventphone-5849@fs:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.19-10-1~64bit 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: 129 Remote-Party-ID: "5849" <sip:5849@fs>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1434120229 1434120230 IN IP4 fs s=FreeSWITCH c=IN IP4 fs t=0 0 m=audio 0 RTP/AVP 19 Freeswitch has a problem establishing the autio connection. It does not send a proper SDP packet back to HG. It does not include a port to send audio to (this is the 0 after m=audio) and also no protocol. Wireshark translates the 19 after RTP/AVP to"Media Format: Comfort noise (old)". The only way to make a proper call between those two is to forbid crypto for incoming POTS calls via HG, which is also not quite good. I think is is a bug in freeswitch, paired with improper behaviour of HGs yate. Normal deskphones seem to work properly when registered to HG and receiving an inbound POTS call, at least the grandstream and snom I have access to. Eventphone does not seem to get complaints from other people registering with them (although most of them use phones and not a PBX). Does anyone know how to debug this further? Thanks Nico

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Re: [Freeswitch-users] Freeswitch 1.6 video recording issue using uuid_record
by Seven Du 21 Aug '15

21 Aug '15

maybe you have a SPACE in your file name? try replace the file name with /tmp/test.mp4 and see if it works. Also, don't send to me directly as it's very hard to REPLY to the list. On Fri, Aug 21, 2015 at 6:05 PM, Ankitt Sharma <akkishrm0825(a)gmail.com> wrote: > hey seven, > We are using freeswitch as media server in integration with kamailio. So > the purpose of using freeswitch is to record the two-way video calls > between the legs > > so without any recording freeswich application or API being used my video > and audio is working just perfect (running on default media handling mode > i.e. proxy_media & bypass media not used), late-negotiation is disabled, > zrtp-passthru is set to false and disable-transcoding is set to true > > after exploring all over web over my recording concern i got to know about > the following APP/APIs: > record, record_session, record_fsv, record_mp4, uuid_record (however i > want to record in .mp4 format) > > so following the very first step I looked for "video-media-bug" branch and > got to know it is merged in freeSWITCH 1.6 master, so used the 1.6 master , > and now the vide-media-bug branch is contained within my current repo > > now using all above mentioned applications i'am not getting my recording > done, here i'am sharing what's happening with uuid_record being used as > > my kbridge dialplan is: > <extension name="kbridge"> > <condition field="destination_number" expression="^kb-(.+)$"> > <action application="set" data="proxy_media=false"/> > <action application="set" data="call_timeout=30"/> > <action application="set" data="continue_on_fail=true"/> > <action application="set" data="hangup_after_bridge=true"/> > <action application="set" data="sip_invite_domain=192.168.5.99"/> > <action application="set" > data="voicemail_authorized=${sip_authorized}"/> > <action application="export" data="sip_contact_user=ufs"/> > > <action application="export" data="api_args=${uuid_setvar ${uuid} > enable_file_write_buffering=false"/> > <action application="export" data="api_result=${uuid_record > ${uuid} start > /usr/local/freeswitch/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_$1_${caller_id_number}_${destination_number}.mp4}"/> > <action application="bridge" data="sofia/$${domain}/$ > 1(a)192.168.5.99"/> > <action application="answer"/> > <action application="voicemail" data="check auth > $${voicemail_profile} default $${domain} $1"/> > </condition> > </extension> > > > and the fs_cli log says: > 2015-08-21 13:19:50.980274 [NOTICE] switch_channel.c:1089 New Channel > sofia/internal/102(a)192.168.5.99 [c1d25c8d-4e6e-458c-abe0-c40b89bc3d68] > 2015-08-21 13:19:50.990233 [INFO] mod_dialplan_xml.c:637 Processing 102 > <102>->kb-101 in context public > 2015-08-21 13:19:50.990233 [NOTICE] switch_ivr.c:1877 Transfer > sofia/internal/102(a)192.168.5.99 to XML[kb-101@default] > 2015-08-21 13:19:50.990233 [INFO] mod_dialplan_xml.c:637 Processing 102 > <102>->kb-101 in context default > 2015-08-21 13:19:51.000223 [INFO] switch_ivr_async.c:2339 Sending early > media > 2015-08-21 13:19:51.000223 [INFO] switch_core_media.c:6543 Activating > VIDEO RTCP PORT 5244 interval 2000 mux 1 > 2015-08-21 13:19:51.000223 [NOTICE] mod_sofia.c:2296 Pre-Answer > sofia/internal/102(a)192.168.5.99! > 2015-08-21 13:19:51.010223 [INFO] mod_native_file.c:101 Opening File > [/usr/local/freeswitch/recordings/2015-08-21-13-19-51-in.G722] 16000hz > 2015-08-21 13:19:51.010223 [INFO] mod_native_file.c:101 Opening File > [/usr/local/freeswitch/recordings/2015-08-21-13-19-51-out.G722] 16000hz > 2015-08-21 13:19:51.010223 [NOTICE] switch_channel.c:1089 New Channel > sofia/internal/101(a)192.168.5.99 [5127cf3a-c7ec-4038-a033-f287b2999a79] > 2015-08-21 13:19:51.220289 [NOTICE] sofia.c:6817 Ring-Ready sofia/internal/ > 101(a)192.168.5.99! > 2015-08-21 13:19:55.240230 [INFO] switch_core_media.c:6543 Activating > VIDEO RTCP PORT 0 interval 2000 mux 0 > 2015-08-21 13:19:55.240230 [NOTICE] sofia.c:7576 Channel [sofia/internal/ > 101(a)192.168.5.99] has been answered > 2015-08-21 13:19:55.240230 [NOTICE] switch_ivr_originate.c:3526 Channel > [sofia/internal/102(a)192.168.5.99] has been answered > 2015-08-21 13:20:04.180275 [NOTICE] sofia.c:952 Hangup sofia/internal/ > 102(a)192.168.5.99 [CS_EXECUTE] [NORMAL_CLEARING] > 2015-08-21 13:20:04.290279 [NOTICE] switch_ivr_bridge.c:874 Hangup > sofia/internal/101(a)192.168.5.99 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2015-08-21 13:20:04.290279 [NOTICE] switch_core_session.c:1657 Session 29 > (sofia/internal/101(a)192.168.5.99) Ended > 2015-08-21 13:20:04.290279 [NOTICE] switch_core_session.c:1661 Close > Channel sofia/internal/101(a)192.168.5.99 [CS_DESTROY] > 2015-08-21 13:20:04.300232 [NOTICE] switch_core_session.c:1657 Session 28 > (sofia/internal/102(a)192.168.5.99) Ended > 2015-08-21 13:20:04.300232 [NOTICE] switch_core_session.c:1661 Close > Channel sofia/internal/102(a)192.168.5.99 [CS_DESTROY] > > > now in the recordings directory i got 2 files(can be seen in the log ) the > files hav the extension .G722, which is not supposed to be the required > output > > since uuid_record is an API I tried using uuid_record in console > so started the call (after commenting the record api in dialplan) and it > gave me this error: > -ERR Cannot locate session! > > > so help me out over this concern > thanks in advance :) > Ankit > -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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